DTMF not working on Slug with Asterisk 1.4 and SlugOSBE

[blmdrew]

Hi.
This is my first post on the forum.
Have been home sick for the last few days and decided it was a good
time to try getting pbx on my Slug.I have a working PBX (Asterisk 1.4)
on another machine so I know that config is good.

My problem is that my voicemail on the Slug is not recognizing any
incoming DTMFs from softphones. (It is recording messages though)

I know that the Softphones are set correctly as they work on the other
PBX on the network.

I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know that
I don't need one that big)

Here is my Sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = others

;register => xxxxxx:xxxxxxx@...:5060
;register => xxxxxx:xxxxxxx@...:5060

register => xxxxxx:xxxxxxx@...:5060
register => xxxxxx:xxxxxxx@...:5060

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic

[2001]
type=friend
context=my-phones
secret=1234
host=dynamic

[link2voip-sw1]
context=from-voip-provider
type=friend
;host=sip.ca1.link2voip.com
host=sip.us1.link2voip.com
username=xxxxxx
secret=xxxxxxx
canreinvite=no ; if using a nat, do not change
insecure=port,invite ; do NOT remove this
qualify=5000 ; do NOT remove this
dtmfmode=auto
nat=no ; do NOT remove/change this
disallow=all
;allow=g729 ;uncomment if you have purchased a g729 license or can do
passthru
allow=ulaw

[link2voip-sw2]
context=from-voip-provider
type=friend                                                          
;host=sip.ca2.link2voip.com
host=sip.us2.link2voip.com
username=xxxxxx
secret=xxxxxxx
canreinvite=no ; if using a nat, do not change
insecure=port,invite ; do NOT remove this
qualify=5000 ; do NOT remove this
dtmfmode=auto
nat=no ; do NOT remove/change this
disallow=all
;allow=g729 ;uncomment if you have purchased a g729 license or can do
passthru
allow=ulaw

I have been scouring posts and googleing for 2 days to no avail.
Any help would be greatly appreciated

Mike

[Ovidiu Sas-3]

try dtmfmode=rfc2833


On Sun, Oct 19, 2008 at 8:00 AM, blmdrew <blmdrew@...> wrote:

> Hi.
> This is my first post on the forum.
> Have been home sick for the last few days and decided it was a good
> time to try getting pbx on my Slug.I have a working PBX (Asterisk 1.4)
> on another machine so I know that config is good.
>
> My problem is that my voicemail on the Slug is not recognizing any
> incoming DTMFs from softphones. (It is recording messages though)
>
> I know that the Softphones are set correctly as they work on the other
> PBX on the network.
>
> I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know that
> I don't need one that big)
>
> Here is my Sip.conf
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = others
>
> ;register => xxxxxx:xxxxxxx@...:5060
> ;register => xxxxxx:xxxxxxx@...:5060
>
> register => xxxxxx:xxxxxxx@...:5060
> register => xxxxxx:xxxxxxx@...:5060
>
> [2000]
> type=friend
> context=my-phones
> secret=1234
> host=dynamic
>
> [2001]
> type=friend
> context=my-phones
> secret=1234
> host=dynamic
>
> [link2voip-sw1]
> context=from-voip-provider
> type=friend
> ;host=sip.ca1.link2voip.com
> host=sip.us1.link2voip.com
> username=xxxxxx
> secret=xxxxxxx
> canreinvite=no ; if using a nat, do not change
> insecure=port,invite ; do NOT remove this
> qualify=5000 ; do NOT remove this
> dtmfmode=auto
> nat=no ; do NOT remove/change this
> disallow=all
> ;allow=g729 ;uncomment if you have purchased a g729 license or can do
> passthru
> allow=ulaw
>
> [link2voip-sw2]
> context=from-voip-provider
> type=friend
> ;host=sip.ca2.link2voip.com
> host=sip.us2.link2voip.com
> username=xxxxxx
> secret=xxxxxxx
> canreinvite=no ; if using a nat, do not change
> insecure=port,invite ; do NOT remove this
> qualify=5000 ; do NOT remove this
> dtmfmode=auto
> nat=no ; do NOT remove/change this
> disallow=all
> ;allow=g729 ;uncomment if you have purchased a g729 license or can do
> passthru
> allow=ulaw
>
> I have been scouring posts and googleing for 2 days to no avail.
> Any help would be greatly appreciated
>
> Mike
>
>
> ------------------------------------
>
> Yahoo! Groups Links
>
>
>
>

[CORNELIU DOBAN]

Try to add this to your phone configuration (in sip.conf) to see if helps:

dtmfmode=rfc2833

But this should be the default anyway (if no other mode is set in the
[general] section).

Good luck,
Corneliu


--- In nslu2-asterisk@..., "blmdrew" <blmdrew@...> wrote:

>
> Hi.
> This is my first post on the forum.
> Have been home sick for the last few days and decided it was a good
> time to try getting pbx on my Slug.I have a working PBX (Asterisk 1.4)
> on another machine so I know that config is good.
>
> My problem is that my voicemail on the Slug is not recognizing any
> incoming DTMFs from softphones. (It is recording messages though)
>
> I know that the Softphones are set correctly as they work on the other
> PBX on the network.
>
> I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know that
> I don't need one that big)
>
> Here is my Sip.conf
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = others
>
> ;register => xxxxxx:xxxxxxx@...:5060
> ;register => xxxxxx:xxxxxxx@...:5060
>
> register => xxxxxx:xxxxxxx@...:5060
> register => xxxxxx:xxxxxxx@...:5060
>
> [2000]
> type=friend
> context=my-phones
> secret=1234
> host=dynamic
>
> [2001]
> type=friend
> context=my-phones
> secret=1234
> host=dynamic
>
> [link2voip-sw1]
> context=from-voip-provider
> type=friend
> ;host=sip.ca1.link2voip.com
> host=sip.us1.link2voip.com
> username=xxxxxx
> secret=xxxxxxx
> canreinvite=no ; if using a nat, do not change
> insecure=port,invite ; do NOT remove this
> qualify=5000 ; do NOT remove this
> dtmfmode=auto
> nat=no ; do NOT remove/change this
> disallow=all
> ;allow=g729 ;uncomment if you have purchased a g729 license or can do
> passthru
> allow=ulaw
>
> [link2voip-sw2]
> context=from-voip-provider
> type=friend                                                          
> ;host=sip.ca2.link2voip.com
> host=sip.us2.link2voip.com
> username=xxxxxx
> secret=xxxxxxx
> canreinvite=no ; if using a nat, do not change
> insecure=port,invite ; do NOT remove this
> qualify=5000 ; do NOT remove this
> dtmfmode=auto
> nat=no ; do NOT remove/change this
> disallow=all
> ;allow=g729 ;uncomment if you have purchased a g729 license or can do
> passthru
> allow=ulaw
>
> I have been scouring posts and googleing for 2 days to no avail.
> Any help would be greatly appreciated
>
> Mike
>

[blmdrew]

Thanks for the quick response.
Yes, I had already tried this dtmfmode=rfc2833 to no avail.
What I did end up doing was going to UnSlung and everything worked fine with
same config.

Than went back to slugosbe and same problem again???

I would much rather go with Slugosbe but not sure why it works with Unslung
& not with slugosbe.


On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
<corneliu_doban@...>wrote:

>
> Try to add this to your phone configuration (in sip.conf) to see if helps:
>
> dtmfmode=rfc2833
>
> But this should be the default anyway (if no other mode is set in the
> [general] section).
>
> Good luck,
> Corneliu
>
>
> --- In nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>,
> "blmdrew" <blmdrew@...> wrote:
> >
> > Hi.
> > This is my first post on the forum.
> > Have been home sick for the last few days and decided it was a good
> > time to try getting pbx on my Slug.I have a working PBX (Asterisk 1.4)
> > on another machine so I know that config is good.
> >
> > My problem is that my voicemail on the Slug is not recognizing any
> > incoming DTMFs from softphones. (It is recording messages though)
> >
> > I know that the Softphones are set correctly as they work on the other
> > PBX on the network.
> >
> > I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know that
> > I don't need one that big)
> >
> > Here is my Sip.conf
> >
> > [general]
> > port = 5060
> > bindaddr = 0.0.0.0
> > context = others
> >
> > ;register => xxxxxx:xxxxxxx@...:5060
> > ;register => xxxxxx:xxxxxxx@...:5060
> >
> > register => xxxxxx:xxxxxxx@...:5060
> > register => xxxxxx:xxxxxxx@...:5060
>
> >
> > [2000]
> > type=friend
> > context=my-phones
> > secret=1234
> > host=dynamic
> >
> > [2001]
> > type=friend
> > context=my-phones
> > secret=1234
> > host=dynamic
> >
> > [link2voip-sw1]
> > context=from-voip-provider
> > type=friend
> > ;host=sip.ca1.link2voip.com
> > host=sip.us1.link2voip.com
> > username=xxxxxx
> > secret=xxxxxxx
> > canreinvite=no ; if using a nat, do not change
> > insecure=port,invite ; do NOT remove this
> > qualify=5000 ; do NOT remove this
> > dtmfmode=auto
> > nat=no ; do NOT remove/change this
> > disallow=all
> > ;allow=g729 ;uncomment if you have purchased a g729 license or can do
> > passthru
> > allow=ulaw
> >
> > [link2voip-sw2]
> > context=from-voip-provider
> > type=friend
> > ;host=sip.ca2.link2voip.com
> > host=sip.us2.link2voip.com
> > username=xxxxxx
> > secret=xxxxxxx
> > canreinvite=no ; if using a nat, do not change
> > insecure=port,invite ; do NOT remove this
> > qualify=5000 ; do NOT remove this
> > dtmfmode=auto
> > nat=no ; do NOT remove/change this
> > disallow=all
> > ;allow=g729 ;uncomment if you have purchased a g729 license or can do
> > passthru
> > allow=ulaw
> >
> > I have been scouring posts and googleing for 2 days to no avail.
> > Any help would be greatly appreciated
> >
> > Mike
> >
>
>  
>

[CORNELIU DOBAN]

I don't think that's something wrong with the slugosbe, although I'm
running a pretty old version: 3.10

You've said that is recording. Are those recordings blanks or you you
have audio?

Corneliu

--- In nslu2-asterisk@..., "Michael Drew" <blmdrew@...> wrote:
>
> Thanks for the quick response.
> Yes, I had already tried this dtmfmode=rfc2833 to no avail.
> What I did end up doing was going to UnSlung and everything worked
fine with
> same config.
>
> Than went back to slugosbe and same problem again???
>
> I would much rather go with Slugosbe but not sure why it works with
Unslung
> & not with slugosbe.
>
>
> On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
> <corneliu_doban@...>wrote:
>
> >
> > Try to add this to your phone configuration (in sip.conf) to see
if helps:

> >
> > dtmfmode=rfc2833
> >
> > But this should be the default anyway (if no other mode is set in the
> > [general] section).
> >
> > Good luck,
> > Corneliu
> >
> >
> > --- In nslu2-asterisk@...

<nslu2-asterisk%40yahoogroups.com>,
> > "blmdrew" <blmdrew@> wrote:
> > >
> > > Hi.
> > > This is my first post on the forum.
> > > Have been home sick for the last few days and decided it was a good
> > > time to try getting pbx on my Slug.I have a working PBX
(Asterisk 1.4)
> > > on another machine so I know that config is good.
> > >
> > > My problem is that my voicemail on the Slug is not recognizing any
> > > incoming DTMFs from softphones. (It is recording messages though)
> > >
> > > I know that the Softphones are set correctly as they work on the
other
> > > PBX on the network.
> > >
> > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know
that

> > > I don't need one that big)
> > >
> > > Here is my Sip.conf
> > >
> > > [general]
> > > port = 5060
> > > bindaddr = 0.0.0.0
> > > context = others
> > >
> > > ;register => xxxxxx:xxxxxxx@:5060
> > > ;register => xxxxxx:xxxxxxx@:5060
> > >
> > > register => xxxxxx:xxxxxxx@:5060
> > > register => xxxxxx:xxxxxxx@:5060
> >
> > >
> > > [2000]
> > > type=friend
> > > context=my-phones
> > > secret=1234
> > > host=dynamic
> > >
> > > [2001]
> > > type=friend
> > > context=my-phones
> > > secret=1234
> > > host=dynamic
> > >
> > > [link2voip-sw1]
> > > context=from-voip-provider
> > > type=friend
> > > ;host=sip.ca1.link2voip.com
> > > host=sip.us1.link2voip.com
> > > username=xxxxxx
> > > secret=xxxxxxx
> > > canreinvite=no ; if using a nat, do not change
> > > insecure=port,invite ; do NOT remove this
> > > qualify=5000 ; do NOT remove this
> > > dtmfmode=auto
> > > nat=no ; do NOT remove/change this
> > > disallow=all
> > > ;allow=g729 ;uncomment if you have purchased a g729 license or

can do

> > > passthru
> > > allow=ulaw
> > >
> > > [link2voip-sw2]
> > > context=from-voip-provider
> > > type=friend
> > > ;host=sip.ca2.link2voip.com
> > > host=sip.us2.link2voip.com
> > > username=xxxxxx
> > > secret=xxxxxxx
> > > canreinvite=no ; if using a nat, do not change
> > > insecure=port,invite ; do NOT remove this
> > > qualify=5000 ; do NOT remove this
> > > dtmfmode=auto
> > > nat=no ; do NOT remove/change this
> > > disallow=all
> > > ;allow=g729 ;uncomment if you have purchased a g729 license or

can do

> > > passthru
> > > allow=ulaw
> > >
> > > I have been scouring posts and googleing for 2 days to no avail.
> > > Any help would be greatly appreciated
> > >
> > > Mike
> > >
> >
> >  
> >
>

[blmdrew]

I do have audio in the recordings i.e. my voice

On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@...>wrote:

>   I don't think that's something wrong with the slugosbe, although I'm
> running a pretty old version: 3.10
>
> You've said that is recording. Are those recordings blanks or you you
> have audio?
>
> Corneliu
>
>
> --- In nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>,
> "Michael Drew" <blmdrew@...> wrote:
> >
> > Thanks for the quick response.
> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
> > What I did end up doing was going to UnSlung and everything worked
> fine with
> > same config.
> >
> > Than went back to slugosbe and same problem again???
> >
> > I would much rather go with Slugosbe but not sure why it works with
> Unslung
> > & not with slugosbe.
> >
> >
> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
> > <corneliu_doban@...>wrote:
> >
> > >
> > > Try to add this to your phone configuration (in sip.conf) to see
> if helps:
> > >
> > > dtmfmode=rfc2833
> > >
> > > But this should be the default anyway (if no other mode is set in the
> > > [general] section).
> > >
> > > Good luck,
> > > Corneliu
> > >
> > >
> > > --- In nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
> <nslu2-asterisk%40yahoogroups.com>,
>
> > > "blmdrew" <blmdrew@> wrote:
> > > >
> > > > Hi.
> > > > This is my first post on the forum.
> > > > Have been home sick for the last few days and decided it was a good
> > > > time to try getting pbx on my Slug.I have a working PBX
> (Asterisk 1.4)
> > > > on another machine so I know that config is good.
> > > >
> > > > My problem is that my voicemail on the Slug is not recognizing any
> > > > incoming DTMFs from softphones. (It is recording messages though)
> > > >
> > > > I know that the Softphones are set correctly as they work on the
> other
> > > > PBX on the network.
> > > >
> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know
> that
> > > > I don't need one that big)
> > > >
> > > > Here is my Sip.conf
> > > >
> > > > [general]
> > > > port = 5060
> > > > bindaddr = 0.0.0.0
> > > > context = others
> > > >
> > > > ;register => xxxxxx:xxxxxxx@:5060
> > > > ;register => xxxxxx:xxxxxxx@:5060
> > > >
> > > > register => xxxxxx:xxxxxxx@:5060
> > > > register => xxxxxx:xxxxxxx@:5060
> > >
> > > >
> > > > [2000]
> > > > type=friend
> > > > context=my-phones
> > > > secret=1234
> > > > host=dynamic
> > > >
> > > > [2001]
> > > > type=friend
> > > > context=my-phones
> > > > secret=1234
> > > > host=dynamic
> > > >
> > > > [link2voip-sw1]
> > > > context=from-voip-provider
> > > > type=friend
> > > > ;host=sip.ca1.link2voip.com
> > > > host=sip.us1.link2voip.com
> > > > username=xxxxxx
> > > > secret=xxxxxxx
> > > > canreinvite=no ; if using a nat, do not change
> > > > insecure=port,invite ; do NOT remove this
> > > > qualify=5000 ; do NOT remove this
> > > > dtmfmode=auto
> > > > nat=no ; do NOT remove/change this
> > > > disallow=all
> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
> can do
> > > > passthru
> > > > allow=ulaw
> > > >
> > > > [link2voip-sw2]
> > > > context=from-voip-provider
> > > > type=friend
> > > > ;host=sip.ca2.link2voip.com
> > > > host=sip.us2.link2voip.com
> > > > username=xxxxxx
> > > > secret=xxxxxxx
> > > > canreinvite=no ; if using a nat, do not change
> > > > insecure=port,invite ; do NOT remove this
> > > > qualify=5000 ; do NOT remove this
> > > > dtmfmode=auto
> > > > nat=no ; do NOT remove/change this
> > > > disallow=all
> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
> can do
> > > > passthru
> > > > allow=ulaw
> > > >
> > > > I have been scouring posts and googleing for 2 days to no avail.
> > > > Any help would be greatly appreciated
> > > >
> > > > Mike
> > > >
> > >
> > >
> > >
> >
>
>  
>

[blmdrew]

I think that I will rebuild with Ver 3.10 to prove or disprove.
I did try again tonight with latest version. I thought that maybe I was
missing something but still same problem
Will keep you posted.

Thanks
Mike

On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@...> wrote:

> I do have audio in the recordings i.e. my voice
>
>
> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@...>wrote:
>
>>   I don't think that's something wrong with the slugosbe, although I'm
>> running a pretty old version: 3.10
>>
>> You've said that is recording. Are those recordings blanks or you you
>> have audio?
>>
>> Corneliu
>>
>>
>> --- In nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>,
>> "Michael Drew" <blmdrew@...> wrote:
>> >
>> > Thanks for the quick response.
>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
>> > What I did end up doing was going to UnSlung and everything worked
>> fine with
>> > same config.
>> >
>> > Than went back to slugosbe and same problem again???
>> >
>> > I would much rather go with Slugosbe but not sure why it works with
>> Unslung
>> > & not with slugosbe.
>> >
>> >
>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
>> > <corneliu_doban@...>wrote:
>> >
>> > >
>> > > Try to add this to your phone configuration (in sip.conf) to see
>> if helps:
>> > >
>> > > dtmfmode=rfc2833
>> > >
>> > > But this should be the default anyway (if no other mode is set in the
>> > > [general] section).
>> > >
>> > > Good luck,
>> > > Corneliu
>> > >
>> > >
>> > > --- In nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
>> <nslu2-asterisk%40yahoogroups.com>,
>>
>> > > "blmdrew" <blmdrew@> wrote:
>> > > >
>> > > > Hi.
>> > > > This is my first post on the forum.
>> > > > Have been home sick for the last few days and decided it was a good
>> > > > time to try getting pbx on my Slug.I have a working PBX
>> (Asterisk 1.4)
>> > > > on another machine so I know that config is good.
>> > > >
>> > > > My problem is that my voicemail on the Slug is not recognizing any
>> > > > incoming DTMFs from softphones. (It is recording messages though)
>> > > >
>> > > > I know that the Softphones are set correctly as they work on the
>> other
>> > > > PBX on the network.
>> > > >
>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know
>> that
>> > > > I don't need one that big)
>> > > >
>> > > > Here is my Sip.conf
>> > > >
>> > > > [general]
>> > > > port = 5060
>> > > > bindaddr = 0.0.0.0
>> > > > context = others
>> > > >
>> > > > ;register => xxxxxx:xxxxxxx@:5060
>> > > > ;register => xxxxxx:xxxxxxx@:5060
>> > > >
>> > > > register => xxxxxx:xxxxxxx@:5060
>> > > > register => xxxxxx:xxxxxxx@:5060
>> > >
>> > > >
>> > > > [2000]
>> > > > type=friend
>> > > > context=my-phones
>> > > > secret=1234
>> > > > host=dynamic
>> > > >
>> > > > [2001]
>> > > > type=friend
>> > > > context=my-phones
>> > > > secret=1234
>> > > > host=dynamic
>> > > >
>> > > > [link2voip-sw1]
>> > > > context=from-voip-provider
>> > > > type=friend
>> > > > ;host=sip.ca1.link2voip.com
>> > > > host=sip.us1.link2voip.com
>> > > > username=xxxxxx
>> > > > secret=xxxxxxx
>> > > > canreinvite=no ; if using a nat, do not change
>> > > > insecure=port,invite ; do NOT remove this
>> > > > qualify=5000 ; do NOT remove this
>> > > > dtmfmode=auto
>> > > > nat=no ; do NOT remove/change this
>> > > > disallow=all
>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
>> can do
>> > > > passthru
>> > > > allow=ulaw
>> > > >
>> > > > [link2voip-sw2]
>> > > > context=from-voip-provider
>> > > > type=friend
>> > > > ;host=sip.ca2.link2voip.com
>> > > > host=sip.us2.link2voip.com
>> > > > username=xxxxxx
>> > > > secret=xxxxxxx
>> > > > canreinvite=no ; if using a nat, do not change
>> > > > insecure=port,invite ; do NOT remove this
>> > > > qualify=5000 ; do NOT remove this
>> > > > dtmfmode=auto
>> > > > nat=no ; do NOT remove/change this
>> > > > disallow=all
>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
>> can do
>> > > > passthru
>> > > > allow=ulaw
>> > > >
>> > > > I have been scouring posts and googleing for 2 days to no avail.
>> > > > Any help would be greatly appreciated
>> > > >
>> > > > Mike
>> > > >
>> > >
>> > >
>> > >
>> >
>>
>>  
>>
>
>

[blmdrew]

I Rebuilt with Ver 3.10 and Asterisk 4.16 and still same problem, so as it
stands:
problem exists when I use - latest slugosbe and Asterisk V14.22
                                            OR:  Ver3.10 with Asterisk
V14.16

It *does work* with Unslung (latest version) and Asterisk V14.22

So I am not sure why there is a problem with OpenSlug and not with
Unslung?????
I can't see anything in the codec/conf files that would cause this.

Just about ready to give in to Unslung

Mike


On Mon, Oct 20, 2008 at 8:11 PM, Michael Drew <blmdrew@...> wrote:

> I think that I will rebuild with Ver 3.10 to prove or disprove.
> I did try again tonight with latest version. I thought that maybe I was
> missing something but still same problem
> Will keep you posted.
>
> Thanks
> Mike
>
> On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@...> wrote:
>
>> I do have audio in the recordings i.e. my voice
>>
>>
>> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@...
>> > wrote:
>>
>>>   I don't think that's something wrong with the slugosbe, although I'm
>>> running a pretty old version: 3.10
>>>
>>> You've said that is recording. Are those recordings blanks or you you
>>> have audio?
>>>
>>> Corneliu
>>>
>>>
>>> --- In nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>,
>>> "Michael Drew" <blmdrew@...> wrote:
>>> >
>>> > Thanks for the quick response.
>>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
>>> > What I did end up doing was going to UnSlung and everything worked
>>> fine with
>>> > same config.
>>> >
>>> > Than went back to slugosbe and same problem again???
>>> >
>>> > I would much rather go with Slugosbe but not sure why it works with
>>> Unslung
>>> > & not with slugosbe.
>>> >
>>> >
>>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
>>> > <corneliu_doban@...>wrote:
>>> >
>>> > >
>>> > > Try to add this to your phone configuration (in sip.conf) to see
>>> if helps:
>>> > >
>>> > > dtmfmode=rfc2833
>>> > >
>>> > > But this should be the default anyway (if no other mode is set in the
>>> > > [general] section).
>>> > >
>>> > > Good luck,
>>> > > Corneliu
>>> > >
>>> > >
>>> > > --- In nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
>>> <nslu2-asterisk%40yahoogroups.com>,
>>>
>>> > > "blmdrew" <blmdrew@> wrote:
>>> > > >
>>> > > > Hi.
>>> > > > This is my first post on the forum.
>>> > > > Have been home sick for the last few days and decided it was a good
>>> > > > time to try getting pbx on my Slug.I have a working PBX
>>> (Asterisk 1.4)
>>> > > > on another machine so I know that config is good.
>>> > > >
>>> > > > My problem is that my voicemail on the Slug is not recognizing any
>>> > > > incoming DTMFs from softphones. (It is recording messages though)
>>> > > >
>>> > > > I know that the Softphones are set correctly as they work on the
>>> other
>>> > > > PBX on the network.
>>> > > >
>>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know
>>> that
>>> > > > I don't need one that big)
>>> > > >
>>> > > > Here is my Sip.conf
>>> > > >
>>> > > > [general]
>>> > > > port = 5060
>>> > > > bindaddr = 0.0.0.0
>>> > > > context = others
>>> > > >
>>> > > > ;register => xxxxxx:xxxxxxx@:5060
>>> > > > ;register => xxxxxx:xxxxxxx@:5060
>>> > > >
>>> > > > register => xxxxxx:xxxxxxx@:5060
>>> > > > register => xxxxxx:xxxxxxx@:5060
>>> > >
>>> > > >
>>> > > > [2000]
>>> > > > type=friend
>>> > > > context=my-phones
>>> > > > secret=1234
>>> > > > host=dynamic
>>> > > >
>>> > > > [2001]
>>> > > > type=friend
>>> > > > context=my-phones
>>> > > > secret=1234
>>> > > > host=dynamic
>>> > > >
>>> > > > [link2voip-sw1]
>>> > > > context=from-voip-provider
>>> > > > type=friend
>>> > > > ;host=sip.ca1.link2voip.com
>>> > > > host=sip.us1.link2voip.com
>>> > > > username=xxxxxx
>>> > > > secret=xxxxxxx
>>> > > > canreinvite=no ; if using a nat, do not change
>>> > > > insecure=port,invite ; do NOT remove this
>>> > > > qualify=5000 ; do NOT remove this
>>> > > > dtmfmode=auto
>>> > > > nat=no ; do NOT remove/change this
>>> > > > disallow=all
>>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
>>> can do
>>> > > > passthru
>>> > > > allow=ulaw
>>> > > >
>>> > > > [link2voip-sw2]
>>> > > > context=from-voip-provider
>>> > > > type=friend
>>> > > > ;host=sip.ca2.link2voip.com
>>> > > > host=sip.us2.link2voip.com
>>> > > > username=xxxxxx
>>> > > > secret=xxxxxxx
>>> > > > canreinvite=no ; if using a nat, do not change
>>> > > > insecure=port,invite ; do NOT remove this
>>> > > > qualify=5000 ; do NOT remove this
>>> > > > dtmfmode=auto
>>> > > > nat=no ; do NOT remove/change this
>>> > > > disallow=all
>>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
>>> can do
>>> > > > passthru
>>> > > > allow=ulaw
>>> > > >
>>> > > > I have been scouring posts and googleing for 2 days to no avail.
>>> > > > Any help would be greatly appreciated
>>> > > >
>>> > > > Mike
>>> > > >
>>> > >
>>> > >
>>> > >
>>> >
>>>
>>>  
>>>
>>
>>
>

[Ovidiu Sas-3]

Have you tried the optware distribution on slugosbe?

On Mon, Oct 20, 2008 at 10:00 PM, Michael Drew <blmdrew@...> wrote:

> I Rebuilt with Ver 3.10 and Asterisk 4.16 and still same problem, so as it
> stands:
> problem exists when I use - latest slugosbe and Asterisk V14.22
>                                             OR:  Ver3.10 with Asterisk
> V14.16
>
> It does work with Unslung (latest version) and Asterisk V14.22
>
> So I am not sure why there is a problem with OpenSlug and not with
> Unslung?????
> I can't see anything in the codec/conf files that would cause this.
>
> Just about ready to give in to Unslung
>
> Mike
>
>
> On Mon, Oct 20, 2008 at 8:11 PM, Michael Drew <blmdrew@...> wrote:
>>
>> I think that I will rebuild with Ver 3.10 to prove or disprove.
>> I did try again tonight with latest version. I thought that maybe I was
>> missing something but still same problem
>> Will keep you posted.
>>
>> Thanks
>> Mike
>>
>> On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@...> wrote:
>>>
>>> I do have audio in the recordings i.e. my voice
>>>
>>> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban
>>> <corneliu_doban@...> wrote:
>>>>
>>>> I don't think that's something wrong with the slugosbe, although I'm
>>>> running a pretty old version: 3.10
>>>>
>>>> You've said that is recording. Are those recordings blanks or you you
>>>> have audio?
>>>>
>>>> Corneliu
>>>>
>>>> --- In nslu2-asterisk@..., "Michael Drew" <blmdrew@...>
>>>> wrote:
>>>> >
>>>> > Thanks for the quick response.
>>>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
>>>> > What I did end up doing was going to UnSlung and everything worked
>>>> fine with
>>>> > same config.
>>>> >
>>>> > Than went back to slugosbe and same problem again???
>>>> >
>>>> > I would much rather go with Slugosbe but not sure why it works with
>>>> Unslung
>>>> > & not with slugosbe.
>>>> >
>>>> >
>>>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
>>>> > <corneliu_doban@...>wrote:
>>>> >
>>>> > >
>>>> > > Try to add this to your phone configuration (in sip.conf) to see
>>>> if helps:
>>>> > >
>>>> > > dtmfmode=rfc2833
>>>> > >
>>>> > > But this should be the default anyway (if no other mode is set in
>>>> > > the
>>>> > > [general] section).
>>>> > >
>>>> > > Good luck,
>>>> > > Corneliu
>>>> > >
>>>> > >
>>>> > > --- In nslu2-asterisk@...
>>>> <nslu2-asterisk%40yahoogroups.com>,
>>>> > > "blmdrew" <blmdrew@> wrote:
>>>> > > >
>>>> > > > Hi.
>>>> > > > This is my first post on the forum.
>>>> > > > Have been home sick for the last few days and decided it was a
>>>> > > > good
>>>> > > > time to try getting pbx on my Slug.I have a working PBX
>>>> (Asterisk 1.4)
>>>> > > > on another machine so I know that config is good.
>>>> > > >
>>>> > > > My problem is that my voicemail on the Slug is not recognizing any
>>>> > > > incoming DTMFs from softphones. (It is recording messages though)
>>>> > > >
>>>> > > > I know that the Softphones are set correctly as they work on the
>>>> other
>>>> > > > PBX on the network.
>>>> > > >
>>>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash drive(I know
>>>> that
>>>> > > > I don't need one that big)
>>>> > > >
>>>> > > > Here is my Sip.conf
>>>> > > >
>>>> > > > [general]
>>>> > > > port = 5060
>>>> > > > bindaddr = 0.0.0.0
>>>> > > > context = others
>>>> > > >
>>>> > > > ;register => xxxxxx:xxxxxxx@:5060
>>>> > > > ;register => xxxxxx:xxxxxxx@:5060
>>>> > > >
>>>> > > > register => xxxxxx:xxxxxxx@:5060
>>>> > > > register => xxxxxx:xxxxxxx@:5060
>>>> > >
>>>> > > >
>>>> > > > [2000]
>>>> > > > type=friend
>>>> > > > context=my-phones
>>>> > > > secret=1234
>>>> > > > host=dynamic
>>>> > > >
>>>> > > > [2001]
>>>> > > > type=friend
>>>> > > > context=my-phones
>>>> > > > secret=1234
>>>> > > > host=dynamic
>>>> > > >
>>>> > > > [link2voip-sw1]
>>>> > > > context=from-voip-provider
>>>> > > > type=friend
>>>> > > > ;host=sip.ca1.link2voip.com
>>>> > > > host=sip.us1.link2voip.com
>>>> > > > username=xxxxxx
>>>> > > > secret=xxxxxxx
>>>> > > > canreinvite=no ; if using a nat, do not change
>>>> > > > insecure=port,invite ; do NOT remove this
>>>> > > > qualify=5000 ; do NOT remove this
>>>> > > > dtmfmode=auto
>>>> > > > nat=no ; do NOT remove/change this
>>>> > > > disallow=all
>>>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
>>>> can do
>>>> > > > passthru
>>>> > > > allow=ulaw
>>>> > > >
>>>> > > > [link2voip-sw2]
>>>> > > > context=from-voip-provider
>>>> > > > type=friend
>>>> > > > ;host=sip.ca2.link2voip.com
>>>> > > > host=sip.us2.link2voip.com
>>>> > > > username=xxxxxx
>>>> > > > secret=xxxxxxx
>>>> > > > canreinvite=no ; if using a nat, do not change
>>>> > > > insecure=port,invite ; do NOT remove this
>>>> > > > qualify=5000 ; do NOT remove this
>>>> > > > dtmfmode=auto
>>>> > > > nat=no ; do NOT remove/change this
>>>> > > > disallow=all
>>>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
>>>> can do
>>>> > > > passthru
>>>> > > > allow=ulaw
>>>> > > >
>>>> > > > I have been scouring posts and googleing for 2 days to no avail.
>>>> > > > Any help would be greatly appreciated
>>>> > > >
>>>> > > > Mike
>>>> > > >
>>>> > >
>>>> > >
>>>> > >
>>>> >
>>>>
>>>
>>
>
>

[CORNELIU DOBAN]

I'm running openslug 3.10  with asterisk 1.4.18 and it works fine.
The problem sounds familiar but I don't remember what was the fix.

Corneliu

--- In nslu2-asterisk@..., "Michael Drew" <blmdrew@...> wrote:
>
> I Rebuilt with Ver 3.10 and Asterisk 4.16 and still same problem, so
as it

> stands:
> problem exists when I use - latest slugosbe and Asterisk V14.22
>                                             OR:  Ver3.10 with Asterisk
> V14.16
>
> It *does work* with Unslung (latest version) and Asterisk V14.22
>
> So I am not sure why there is a problem with OpenSlug and not with
> Unslung?????
> I can't see anything in the codec/conf files that would cause this.
>
> Just about ready to give in to Unslung
>
> Mike
>
>
> On Mon, Oct 20, 2008 at 8:11 PM, Michael Drew <blmdrew@...> wrote:
>
> > I think that I will rebuild with Ver 3.10 to prove or disprove.
> > I did try again tonight with latest version. I thought that maybe

I was

> > missing something but still same problem
> > Will keep you posted.
> >
> > Thanks
> > Mike
> >
> > On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@...> wrote:
> >
> >> I do have audio in the recordings i.e. my voice
> >>
> >>
> >> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@...
> >> > wrote:
> >>
> >>>   I don't think that's something wrong with the slugosbe,

although I'm
> >>> running a pretty old version: 3.10
> >>>
> >>> You've said that is recording. Are those recordings blanks or
you you
> >>> have audio?
> >>>
> >>> Corneliu
> >>>
> >>>
> >>> --- In nslu2-asterisk@...
<nslu2-asterisk%40yahoogroups.com>,

> >>> "Michael Drew" <blmdrew@> wrote:
> >>> >
> >>> > Thanks for the quick response.
> >>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
> >>> > What I did end up doing was going to UnSlung and everything worked
> >>> fine with
> >>> > same config.
> >>> >
> >>> > Than went back to slugosbe and same problem again???
> >>> >
> >>> > I would much rather go with Slugosbe but not sure why it works

with

> >>> Unslung
> >>> > & not with slugosbe.
> >>> >
> >>> >
> >>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
> >>> > <corneliu_doban@>wrote:
> >>> >
> >>> > >
> >>> > > Try to add this to your phone configuration (in sip.conf) to see
> >>> if helps:
> >>> > >
> >>> > > dtmfmode=rfc2833
> >>> > >
> >>> > > But this should be the default anyway (if no other mode is

set in the
> >>> > > [general] section).
> >>> > >
> >>> > > Good luck,
> >>> > > Corneliu
> >>> > >
> >>> > >
> >>> > > --- In
nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
> >>> <nslu2-asterisk%40yahoogroups.com>,
> >>>
> >>> > > "blmdrew" <blmdrew@> wrote:
> >>> > > >
> >>> > > > Hi.
> >>> > > > This is my first post on the forum.
> >>> > > > Have been home sick for the last few days and decided it
was a good
> >>> > > > time to try getting pbx on my Slug.I have a working PBX
> >>> (Asterisk 1.4)
> >>> > > > on another machine so I know that config is good.
> >>> > > >
> >>> > > > My problem is that my voicemail on the Slug is not
recognizing any
> >>> > > > incoming DTMFs from softphones. (It is recording messages
though)
> >>> > > >
> >>> > > > I know that the Softphones are set correctly as they work
on the
> >>> other
> >>> > > > PBX on the network.
> >>> > > >
> >>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash
drive(I know

> >>> that
> >>> > > > I don't need one that big)
> >>> > > >
> >>> > > > Here is my Sip.conf
> >>> > > >
> >>> > > > [general]
> >>> > > > port = 5060
> >>> > > > bindaddr = 0.0.0.0
> >>> > > > context = others
> >>> > > >
> >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> >>> > > >
> >>> > > > register => xxxxxx:xxxxxxx@:5060
> >>> > > > register => xxxxxx:xxxxxxx@:5060
> >>> > >
> >>> > > >
> >>> > > > [2000]
> >>> > > > type=friend
> >>> > > > context=my-phones
> >>> > > > secret=1234
> >>> > > > host=dynamic
> >>> > > >
> >>> > > > [2001]
> >>> > > > type=friend
> >>> > > > context=my-phones
> >>> > > > secret=1234
> >>> > > > host=dynamic
> >>> > > >
> >>> > > > [link2voip-sw1]
> >>> > > > context=from-voip-provider
> >>> > > > type=friend
> >>> > > > ;host=sip.ca1.link2voip.com
> >>> > > > host=sip.us1.link2voip.com
> >>> > > > username=xxxxxx
> >>> > > > secret=xxxxxxx
> >>> > > > canreinvite=no ; if using a nat, do not change
> >>> > > > insecure=port,invite ; do NOT remove this
> >>> > > > qualify=5000 ; do NOT remove this
> >>> > > > dtmfmode=auto
> >>> > > > nat=no ; do NOT remove/change this
> >>> > > > disallow=all
> >>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
> >>> can do
> >>> > > > passthru
> >>> > > > allow=ulaw
> >>> > > >
> >>> > > > [link2voip-sw2]
> >>> > > > context=from-voip-provider
> >>> > > > type=friend
> >>> > > > ;host=sip.ca2.link2voip.com
> >>> > > > host=sip.us2.link2voip.com
> >>> > > > username=xxxxxx
> >>> > > > secret=xxxxxxx
> >>> > > > canreinvite=no ; if using a nat, do not change
> >>> > > > insecure=port,invite ; do NOT remove this
> >>> > > > qualify=5000 ; do NOT remove this
> >>> > > > dtmfmode=auto
> >>> > > > nat=no ; do NOT remove/change this
> >>> > > > disallow=all
> >>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
> >>> can do
> >>> > > > passthru
> >>> > > > allow=ulaw
> >>> > > >
> >>> > > > I have been scouring posts and googleing for 2 days to no

avail.

> >>> > > > Any help would be greatly appreciated
> >>> > > >
> >>> > > > Mike
> >>> > > >
> >>> > >
> >>> > >
> >>> > >
> >>> >
> >>>
> >>>  
> >>>
> >>
> >>
> >
>

[blmdrew]

Did you encounter the problem when you initially installed 3.10 with 1.4.18?

Just wondering if I should try that combination

I did try the opt packages but same problem.

I went back to unslung today with the optware/nslu2/cross/stable feeds & is
ok .

On Tue, Oct 21, 2008 at 10:41 AM, Corneliu Doban
<corneliu_doban@...>wrote:

>   I'm running openslug 3.10 with asterisk 1.4.18 and it works fine.
> The problem sounds familiar but I don't remember what was the fix.
>
>
> Corneliu
>
> --- In nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>,
> "Michael Drew" <blmdrew@...> wrote:
> >
> > I Rebuilt with Ver 3.10 and Asterisk 4.16 and still same problem, so
> as it
> > stands:
> > problem exists when I use - latest slugosbe and Asterisk V14.22
> > OR: Ver3.10 with Asterisk
> > V14.16
> >
> > It *does work* with Unslung (latest version) and Asterisk V14.22
> >
> > So I am not sure why there is a problem with OpenSlug and not with
> > Unslung?????
> > I can't see anything in the codec/conf files that would cause this.
> >
> > Just about ready to give in to Unslung
> >
> > Mike
> >
> >
> > On Mon, Oct 20, 2008 at 8:11 PM, Michael Drew <blmdrew@...> wrote:
> >
> > > I think that I will rebuild with Ver 3.10 to prove or disprove.
> > > I did try again tonight with latest version. I thought that maybe
> I was
> > > missing something but still same problem
> > > Will keep you posted.
> > >
> > > Thanks
> > > Mike
> > >
> > > On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@...> wrote:
> > >
> > >> I do have audio in the recordings i.e. my voice
> > >>
> > >>
> > >> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@...
> > >> > wrote:
> > >>
> > >>> I don't think that's something wrong with the slugosbe,
> although I'm
> > >>> running a pretty old version: 3.10
> > >>>
> > >>> You've said that is recording. Are those recordings blanks or
> you you
> > >>> have audio?
> > >>>
> > >>> Corneliu
> > >>>
> > >>>
> > >>> --- In nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
> <nslu2-asterisk%40yahoogroups.com>,
>
> > >>> "Michael Drew" <blmdrew@> wrote:
> > >>> >
> > >>> > Thanks for the quick response.
> > >>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
> > >>> > What I did end up doing was going to UnSlung and everything worked
> > >>> fine with
> > >>> > same config.
> > >>> >
> > >>> > Than went back to slugosbe and same problem again???
> > >>> >
> > >>> > I would much rather go with Slugosbe but not sure why it works
> with
> > >>> Unslung
> > >>> > & not with slugosbe.
> > >>> >
> > >>> >
> > >>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
> > >>> > <corneliu_doban@>wrote:
> > >>> >
> > >>> > >
> > >>> > > Try to add this to your phone configuration (in sip.conf) to see
> > >>> if helps:
> > >>> > >
> > >>> > > dtmfmode=rfc2833
> > >>> > >
> > >>> > > But this should be the default anyway (if no other mode is
> set in the
> > >>> > > [general] section).
> > >>> > >
> > >>> > > Good luck,
> > >>> > > Corneliu
> > >>> > >
> > >>> > >
> > >>> > > --- In
> nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>
> <nslu2-asterisk%40yahoogroups.com>
> > >>> <nslu2-asterisk%40yahoogroups.com>,
> > >>>
> > >>> > > "blmdrew" <blmdrew@> wrote:
> > >>> > > >
> > >>> > > > Hi.
> > >>> > > > This is my first post on the forum.
> > >>> > > > Have been home sick for the last few days and decided it
> was a good
> > >>> > > > time to try getting pbx on my Slug.I have a working PBX
> > >>> (Asterisk 1.4)
> > >>> > > > on another machine so I know that config is good.
> > >>> > > >
> > >>> > > > My problem is that my voicemail on the Slug is not
> recognizing any
> > >>> > > > incoming DTMFs from softphones. (It is recording messages
> though)
> > >>> > > >
> > >>> > > > I know that the Softphones are set correctly as they work
> on the
> > >>> other
> > >>> > > > PBX on the network.
> > >>> > > >
> > >>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash
> drive(I know
> > >>> that
> > >>> > > > I don't need one that big)
> > >>> > > >
> > >>> > > > Here is my Sip.conf
> > >>> > > >
> > >>> > > > [general]
> > >>> > > > port = 5060
> > >>> > > > bindaddr = 0.0.0.0
> > >>> > > > context = others
> > >>> > > >
> > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> > >>> > > >
> > >>> > > > register => xxxxxx:xxxxxxx@:5060
> > >>> > > > register => xxxxxx:xxxxxxx@:5060
> > >>> > >
> > >>> > > >
> > >>> > > > [2000]
> > >>> > > > type=friend
> > >>> > > > context=my-phones
> > >>> > > > secret=1234
> > >>> > > > host=dynamic
> > >>> > > >
> > >>> > > > [2001]
> > >>> > > > type=friend
> > >>> > > > context=my-phones
> > >>> > > > secret=1234
> > >>> > > > host=dynamic
> > >>> > > >
> > >>> > > > [link2voip-sw1]
> > >>> > > > context=from-voip-provider
> > >>> > > > type=friend
> > >>> > > > ;host=sip.ca1.link2voip.com
> > >>> > > > host=sip.us1.link2voip.com
> > >>> > > > username=xxxxxx
> > >>> > > > secret=xxxxxxx
> > >>> > > > canreinvite=no ; if using a nat, do not change
> > >>> > > > insecure=port,invite ; do NOT remove this
> > >>> > > > qualify=5000 ; do NOT remove this
> > >>> > > > dtmfmode=auto
> > >>> > > > nat=no ; do NOT remove/change this
> > >>> > > > disallow=all
> > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
> > >>> can do
> > >>> > > > passthru
> > >>> > > > allow=ulaw
> > >>> > > >
> > >>> > > > [link2voip-sw2]
> > >>> > > > context=from-voip-provider
> > >>> > > > type=friend
> > >>> > > > ;host=sip.ca2.link2voip.com
> > >>> > > > host=sip.us2.link2voip.com
> > >>> > > > username=xxxxxx
> > >>> > > > secret=xxxxxxx
> > >>> > > > canreinvite=no ; if using a nat, do not change
> > >>> > > > insecure=port,invite ; do NOT remove this
> > >>> > > > qualify=5000 ; do NOT remove this
> > >>> > > > dtmfmode=auto
> > >>> > > > nat=no ; do NOT remove/change this
> > >>> > > > disallow=all
> > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729 license or
> > >>> can do
> > >>> > > > passthru
> > >>> > > > allow=ulaw
> > >>> > > >
> > >>> > > > I have been scouring posts and googleing for 2 days to no
> avail.
> > >>> > > > Any help would be greatly appreciated
> > >>> > > >
> > >>> > > > Mike
> > >>> > > >
> > >>> > >
> > >>> > >
> > >>> > >
> > >>> >
> > >>>
> > >>>
> > >>>
> > >>
> > >>
> > >
> >
>
>  
>

[CORNELIU DOBAN]

No, initially everything worked fine (but I had asterisk 1.2 when I
installed the Openslug 3.10).
I don't remember when exactly I had this issue.

Do you use the IP address or the FQDN of the asterisk for the
configured SIP proxy and registrar on the softphones?

I was using the FQDN and I remember that I had to change it to the IP
address when I changed the ISP. Not sure but I suspect that this was
also the moment when I encountered your problem too.

Corneliu

--- In nslu2-asterisk@..., "Michael Drew" <blmdrew@...> wrote:
>
> Did you encounter the problem when you initially installed 3.10 with
1.4.18?
>
> Just wondering if I should try that combination
>
> I did try the opt packages but same problem.
>
> I went back to unslung today with the optware/nslu2/cross/stable
feeds & is

> ok .
>
> On Tue, Oct 21, 2008 at 10:41 AM, Corneliu Doban
> <corneliu_doban@...>wrote:
>
> >   I'm running openslug 3.10 with asterisk 1.4.18 and it works fine.
> > The problem sounds familiar but I don't remember what was the fix.
> >
> >
> > Corneliu
> >
> > --- In nslu2-asterisk@...

<nslu2-asterisk%40yahoogroups.com>,

> > "Michael Drew" <blmdrew@> wrote:
> > >
> > > I Rebuilt with Ver 3.10 and Asterisk 4.16 and still same problem, so
> > as it
> > > stands:
> > > problem exists when I use - latest slugosbe and Asterisk V14.22
> > > OR: Ver3.10 with Asterisk
> > > V14.16
> > >
> > > It *does work* with Unslung (latest version) and Asterisk V14.22
> > >
> > > So I am not sure why there is a problem with OpenSlug and not with
> > > Unslung?????
> > > I can't see anything in the codec/conf files that would cause this.
> > >
> > > Just about ready to give in to Unslung
> > >
> > > Mike
> > >
> > >
> > > On Mon, Oct 20, 2008 at 8:11 PM, Michael Drew <blmdrew@> wrote:
> > >
> > > > I think that I will rebuild with Ver 3.10 to prove or disprove.
> > > > I did try again tonight with latest version. I thought that maybe
> > I was
> > > > missing something but still same problem
> > > > Will keep you posted.
> > > >
> > > > Thanks
> > > > Mike
> > > >
> > > > On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@> wrote:
> > > >
> > > >> I do have audio in the recordings i.e. my voice
> > > >>
> > > >>
> > > >> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@
> > > >> > wrote:
> > > >>
> > > >>> I don't think that's something wrong with the slugosbe,
> > although I'm
> > > >>> running a pretty old version: 3.10
> > > >>>
> > > >>> You've said that is recording. Are those recordings blanks or
> > you you
> > > >>> have audio?
> > > >>>
> > > >>> Corneliu
> > > >>>
> > > >>>
> > > >>> --- In

nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
> > <nslu2-asterisk%40yahoogroups.com>,
> >
> > > >>> "Michael Drew" <blmdrew@> wrote:
> > > >>> >
> > > >>> > Thanks for the quick response.
> > > >>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
> > > >>> > What I did end up doing was going to UnSlung and
everything worked

> > > >>> fine with
> > > >>> > same config.
> > > >>> >
> > > >>> > Than went back to slugosbe and same problem again???
> > > >>> >
> > > >>> > I would much rather go with Slugosbe but not sure why it works
> > with
> > > >>> Unslung
> > > >>> > & not with slugosbe.
> > > >>> >
> > > >>> >
> > > >>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
> > > >>> > <corneliu_doban@>wrote:
> > > >>> >
> > > >>> > >
> > > >>> > > Try to add this to your phone configuration (in

sip.conf) to see

> > > >>> if helps:
> > > >>> > >
> > > >>> > > dtmfmode=rfc2833
> > > >>> > >
> > > >>> > > But this should be the default anyway (if no other mode is
> > set in the
> > > >>> > > [general] section).
> > > >>> > >
> > > >>> > > Good luck,
> > > >>> > > Corneliu
> > > >>> > >
> > > >>> > >
> > > >>> > > --- In
> > nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>
> > <nslu2-asterisk%40yahoogroups.com>
> > > >>> <nslu2-asterisk%40yahoogroups.com>,
> > > >>>
> > > >>> > > "blmdrew" <blmdrew@> wrote:
> > > >>> > > >
> > > >>> > > > Hi.
> > > >>> > > > This is my first post on the forum.
> > > >>> > > > Have been home sick for the last few days and decided it
> > was a good
> > > >>> > > > time to try getting pbx on my Slug.I have a working PBX
> > > >>> (Asterisk 1.4)
> > > >>> > > > on another machine so I know that config is good.
> > > >>> > > >
> > > >>> > > > My problem is that my voicemail on the Slug is not
> > recognizing any
> > > >>> > > > incoming DTMFs from softphones. (It is recording messages
> > though)
> > > >>> > > >
> > > >>> > > > I know that the Softphones are set correctly as they work
> > on the
> > > >>> other
> > > >>> > > > PBX on the network.
> > > >>> > > >
> > > >>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash
> > drive(I know
> > > >>> that
> > > >>> > > > I don't need one that big)
> > > >>> > > >
> > > >>> > > > Here is my Sip.conf
> > > >>> > > >
> > > >>> > > > [general]
> > > >>> > > > port = 5060
> > > >>> > > > bindaddr = 0.0.0.0
> > > >>> > > > context = others
> > > >>> > > >
> > > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> > > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> > > >>> > > >
> > > >>> > > > register => xxxxxx:xxxxxxx@:5060
> > > >>> > > > register => xxxxxx:xxxxxxx@:5060
> > > >>> > >
> > > >>> > > >
> > > >>> > > > [2000]
> > > >>> > > > type=friend
> > > >>> > > > context=my-phones
> > > >>> > > > secret=1234
> > > >>> > > > host=dynamic
> > > >>> > > >
> > > >>> > > > [2001]
> > > >>> > > > type=friend
> > > >>> > > > context=my-phones
> > > >>> > > > secret=1234
> > > >>> > > > host=dynamic
> > > >>> > > >
> > > >>> > > > [link2voip-sw1]
> > > >>> > > > context=from-voip-provider
> > > >>> > > > type=friend
> > > >>> > > > ;host=sip.ca1.link2voip.com
> > > >>> > > > host=sip.us1.link2voip.com
> > > >>> > > > username=xxxxxx
> > > >>> > > > secret=xxxxxxx
> > > >>> > > > canreinvite=no ; if using a nat, do not change
> > > >>> > > > insecure=port,invite ; do NOT remove this
> > > >>> > > > qualify=5000 ; do NOT remove this
> > > >>> > > > dtmfmode=auto
> > > >>> > > > nat=no ; do NOT remove/change this
> > > >>> > > > disallow=all
> > > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729

license or

> > > >>> can do
> > > >>> > > > passthru
> > > >>> > > > allow=ulaw
> > > >>> > > >
> > > >>> > > > [link2voip-sw2]
> > > >>> > > > context=from-voip-provider
> > > >>> > > > type=friend
> > > >>> > > > ;host=sip.ca2.link2voip.com
> > > >>> > > > host=sip.us2.link2voip.com
> > > >>> > > > username=xxxxxx
> > > >>> > > > secret=xxxxxxx
> > > >>> > > > canreinvite=no ; if using a nat, do not change
> > > >>> > > > insecure=port,invite ; do NOT remove this
> > > >>> > > > qualify=5000 ; do NOT remove this
> > > >>> > > > dtmfmode=auto
> > > >>> > > > nat=no ; do NOT remove/change this
> > > >>> > > > disallow=all
> > > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729

license or

> > > >>> can do
> > > >>> > > > passthru
> > > >>> > > > allow=ulaw
> > > >>> > > >
> > > >>> > > > I have been scouring posts and googleing for 2 days to no
> > avail.
> > > >>> > > > Any help would be greatly appreciated
> > > >>> > > >
> > > >>> > > > Mike
> > > >>> > > >
> > > >>> > >
> > > >>> > >
> > > >>> > >
> > > >>> >
> > > >>>
> > > >>>
> > > >>>
> > > >>
> > > >>
> > > >
> > >
> >
> >  
> >
>

[blmdrew]

I have a static i.p. address on the Slug of 192.168.1.4 and am accessing
locally from another machine running Ekiga.
Also, I am dialing in from my DID and experiencing the same problem.
I just figured out how to use the Asterisk debugging and here are some
results.
With debugging and rtp debugging set this is what I get when dialing in to
my voicemail at X200 from ekiga:
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
write format ulaw
[Nov  2 15:31:45] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
write format gsm
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] sched.c: Request to schedule in the past?!?!
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
[Nov  2 15:31:46] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
write format ulaw
[Nov  2 15:31:46] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
write format gsm
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
[Nov  2 15:31:52] DEBUG[1065] rtp.c: Got RTCP report of 88 bytes
[Nov  2 15:31:53] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
write format ulaw
[Nov  2 15:31:53] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
write format gsm
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
[Nov  2 15:31:56] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
write format ulaw
[Nov  2 15:31:57] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:57] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:57] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:57] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:57] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:57] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
[Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
192.168.1.147'
[Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)

Notice that Asterisk is detecting DTMF but rejecting it.
You can even tell which keys were pressed  by the event   i.e.    1  3  0  #

When I dial to my DID I get basically the same results so whether I am using
Ekiga or my DID, Asterisk rejects the DTMFs. I re-confirmed that the DTMFs
were working ok on UnSlung. This is using lastest SlugOSBE with Asterisk
1.4.22

Thanks
Mike


On Wed, Oct 22, 2008 at 11:09 AM, Corneliu Doban
<corneliu_doban@...>wrote:

>   No, initially everything worked fine (but I had asterisk 1.2 when I
> installed the Openslug 3.10).
> I don't remember when exactly I had this issue.
>
> Do you use the IP address or the FQDN of the asterisk for the
> configured SIP proxy and registrar on the softphones?
>
> I was using the FQDN and I remember that I had to change it to the IP
> address when I changed the ISP. Not sure but I suspect that this was
> also the moment when I encountered your problem too.
>
>
> Corneliu
>
> --- In nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>,
> "Michael Drew" <blmdrew@...> wrote:
> >
> > Did you encounter the problem when you initially installed 3.10 with
> 1.4.18?
> >
> > Just wondering if I should try that combination
> >
> > I did try the opt packages but same problem.
> >
> > I went back to unslung today with the optware/nslu2/cross/stable
> feeds & is
> > ok .
> >
> > On Tue, Oct 21, 2008 at 10:41 AM, Corneliu Doban
> > <corneliu_doban@...>wrote:
> >
> > > I'm running openslug 3.10 with asterisk 1.4.18 and it works fine.
> > > The problem sounds familiar but I don't remember what was the fix.
> > >
> > >
> > > Corneliu
> > >
> > > --- In nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
> <nslu2-asterisk%40yahoogroups.com>,
> > > "Michael Drew" <blmdrew@> wrote:
> > > >
> > > > I Rebuilt with Ver 3.10 and Asterisk 4.16 and still same problem, so
> > > as it
> > > > stands:
> > > > problem exists when I use - latest slugosbe and Asterisk V14.22
> > > > OR: Ver3.10 with Asterisk
> > > > V14.16
> > > >
> > > > It *does work* with Unslung (latest version) and Asterisk V14.22
> > > >
> > > > So I am not sure why there is a problem with OpenSlug and not with
> > > > Unslung?????
> > > > I can't see anything in the codec/conf files that would cause this.
> > > >
> > > > Just about ready to give in to Unslung
> > > >
> > > > Mike
> > > >
> > > >
> > > > On Mon, Oct 20, 2008 at 8:11 PM, Michael Drew <blmdrew@> wrote:
> > > >
> > > > > I think that I will rebuild with Ver 3.10 to prove or disprove.
> > > > > I did try again tonight with latest version. I thought that maybe
> > > I was
> > > > > missing something but still same problem
> > > > > Will keep you posted.
> > > > >
> > > > > Thanks
> > > > > Mike
> > > > >
> > > > > On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@> wrote:
> > > > >
> > > > >> I do have audio in the recordings i.e. my voice
> > > > >>
> > > > >>
> > > > >> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@
> > > > >> > wrote:
> > > > >>
> > > > >>> I don't think that's something wrong with the slugosbe,
> > > although I'm
> > > > >>> running a pretty old version: 3.10
> > > > >>>
> > > > >>> You've said that is recording. Are those recordings blanks or
> > > you you
> > > > >>> have audio?
> > > > >>>
> > > > >>> Corneliu
> > > > >>>
> > > > >>>
> > > > >>> --- In
> nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>
> <nslu2-asterisk%40yahoogroups.com>
> > > <nslu2-asterisk%40yahoogroups.com>,
> > >
> > > > >>> "Michael Drew" <blmdrew@> wrote:
> > > > >>> >
> > > > >>> > Thanks for the quick response.
> > > > >>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
> > > > >>> > What I did end up doing was going to UnSlung and
> everything worked
> > > > >>> fine with
> > > > >>> > same config.
> > > > >>> >
> > > > >>> > Than went back to slugosbe and same problem again???
> > > > >>> >
> > > > >>> > I would much rather go with Slugosbe but not sure why it works
> > > with
> > > > >>> Unslung
> > > > >>> > & not with slugosbe.
> > > > >>> >
> > > > >>> >
> > > > >>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
> > > > >>> > <corneliu_doban@>wrote:
> > > > >>> >
> > > > >>> > >
> > > > >>> > > Try to add this to your phone configuration (in
> sip.conf) to see
> > > > >>> if helps:
> > > > >>> > >
> > > > >>> > > dtmfmode=rfc2833
> > > > >>> > >
> > > > >>> > > But this should be the default anyway (if no other mode is
> > > set in the
> > > > >>> > > [general] section).
> > > > >>> > >
> > > > >>> > > Good luck,
> > > > >>> > > Corneliu
> > > > >>> > >
> > > > >>> > >
> > > > >>> > > --- In
> > > nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com><nslu2-asterisk%
> 40yahoogroups.com>
> > > <nslu2-asterisk%40yahoogroups.com>
> > > > >>> <nslu2-asterisk%40yahoogroups.com>,
> > > > >>>
> > > > >>> > > "blmdrew" <blmdrew@> wrote:
> > > > >>> > > >
> > > > >>> > > > Hi.
> > > > >>> > > > This is my first post on the forum.
> > > > >>> > > > Have been home sick for the last few days and decided it
> > > was a good
> > > > >>> > > > time to try getting pbx on my Slug.I have a working PBX
> > > > >>> (Asterisk 1.4)
> > > > >>> > > > on another machine so I know that config is good.
> > > > >>> > > >
> > > > >>> > > > My problem is that my voicemail on the Slug is not
> > > recognizing any
> > > > >>> > > > incoming DTMFs from softphones. (It is recording messages
> > > though)
> > > > >>> > > >
> > > > >>> > > > I know that the Softphones are set correctly as they work
> > > on the
> > > > >>> other
> > > > >>> > > > PBX on the network.
> > > > >>> > > >
> > > > >>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash
> > > drive(I know
> > > > >>> that
> > > > >>> > > > I don't need one that big)
> > > > >>> > > >
> > > > >>> > > > Here is my Sip.conf
> > > > >>> > > >
> > > > >>> > > > [general]
> > > > >>> > > > port = 5060
> > > > >>> > > > bindaddr = 0.0.0.0
> > > > >>> > > > context = others
> > > > >>> > > >
> > > > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> > > > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
> > > > >>> > > >
> > > > >>> > > > register => xxxxxx:xxxxxxx@:5060
> > > > >>> > > > register => xxxxxx:xxxxxxx@:5060
> > > > >>> > >
> > > > >>> > > >
> > > > >>> > > > [2000]
> > > > >>> > > > type=friend
> > > > >>> > > > context=my-phones
> > > > >>> > > > secret=1234
> > > > >>> > > > host=dynamic
> > > > >>> > > >
> > > > >>> > > > [2001]
> > > > >>> > > > type=friend
> > > > >>> > > > context=my-phones
> > > > >>> > > > secret=1234
> > > > >>> > > > host=dynamic
> > > > >>> > > >
> > > > >>> > > > [link2voip-sw1]
> > > > >>> > > > context=from-voip-provider
> > > > >>> > > > type=friend
> > > > >>> > > > ;host=sip.ca1.link2voip.com
> > > > >>> > > > host=sip.us1.link2voip.com
> > > > >>> > > > username=xxxxxx
> > > > >>> > > > secret=xxxxxxx
> > > > >>> > > > canreinvite=no ; if using a nat, do not change
> > > > >>> > > > insecure=port,invite ; do NOT remove this
> > > > >>> > > > qualify=5000 ; do NOT remove this
> > > > >>> > > > dtmfmode=auto
> > > > >>> > > > nat=no ; do NOT remove/change this
> > > > >>> > > > disallow=all
> > > > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729
> license or
> > > > >>> can do
> > > > >>> > > > passthru
> > > > >>> > > > allow=ulaw
> > > > >>> > > >
> > > > >>> > > > [link2voip-sw2]
> > > > >>> > > > context=from-voip-provider
> > > > >>> > > > type=friend
> > > > >>> > > > ;host=sip.ca2.link2voip.com
> > > > >>> > > > host=sip.us2.link2voip.com
> > > > >>> > > > username=xxxxxx
> > > > >>> > > > secret=xxxxxxx
> > > > >>> > > > canreinvite=no ; if using a nat, do not change
> > > > >>> > > > insecure=port,invite ; do NOT remove this
> > > > >>> > > > qualify=5000 ; do NOT remove this
> > > > >>> > > > dtmfmode=auto
> > > > >>> > > > nat=no ; do NOT remove/change this
> > > > >>> > > > disallow=all
> > > > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729
> license or
> > > > >>> can do
> > > > >>> > > > passthru
> > > > >>> > > > allow=ulaw
> > > > >>> > > >
> > > > >>> > > > I have been scouring posts and googleing for 2 days to no
> > > avail.
> > > > >>> > > > Any help would be greatly appreciated
> > > > >>> > > >
> > > > >>> > > > Mike
> > > > >>> > > >
> > > > >>> > >
> > > > >>> > >
> > > > >>> > >
> > > > >>> >
> > > > >>>
> > > > >>>
> > > > >>>
> > > > >>
> > > > >>
> > > > >
> > > >
> > >
> > >
> > >
> >
>
>  
>

[blmdrew]

PROBLEM SOLVED!!
I found that in order for the DTMF to be detected in Slugosbe, my Slug clock
have to be set correctly.
After setting up with NTPClient on startup and in Cron, I have no more
issues.
I still don't know why Upslung worked but I wonder if NTPClient was set up
automatically in the UpSlung build??


On Wed, Oct 22, 2008 at 10:20 PM, Michael Drew <blmdrew@...> wrote:

> I have a static i.p. address on the Slug of 192.168.1.4 and am accessing
> locally from another machine running Ekiga.
> Also, I am dialing in from my DID and experiencing the same problem.
> I just figured out how to use the Asterisk debugging and here are some
> results.
> With debugging and rtp debugging set this is what I get when dialing in to
> my voicemail at X200 from ekiga:
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
> write format ulaw
> [Nov  2 15:31:45] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
> write format gsm
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] sched.c: Request to schedule in the past?!?!
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:45] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
> [Nov  2 15:31:46] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
> write format ulaw
> [Nov  2 15:31:46] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
> write format gsm
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:51] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000003 (len = 4)
> [Nov  2 15:31:52] DEBUG[1065] rtp.c: Got RTCP report of 88 bytes
> [Nov  2 15:31:53] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
> write format ulaw
> [Nov  2 15:31:53] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
> write format gsm
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:54] DEBUG[1065] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
> [Nov  2 15:31:56] DEBUG[1065] channel.c: Set channel SIP/2000-001758b0 to
> write format ulaw
> [Nov  2 15:31:57] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:57] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:57] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:57] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:57] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:57] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: Ignore potential DTMF echo from '
> 192.168.1.147'
> [Nov  2 15:31:58] DEBUG[1065] rtp.c: - RTP 2833 Event: 0000000b (len = 4)
>
> Notice that Asterisk is detecting DTMF but rejecting it.
> You can even tell which keys were pressed  by the event   i.e.    1  3  0
> #
>
> When I dial to my DID I get basically the same results so whether I am
> using Ekiga or my DID, Asterisk rejects the DTMFs. I re-confirmed that the
> DTMFs were working ok on UnSlung. This is using lastest SlugOSBE with
> Asterisk 1.4.22
>
> Thanks
> Mike
>
>
>
> On Wed, Oct 22, 2008 at 11:09 AM, Corneliu Doban <corneliu_doban@...
> > wrote:
>
>>   No, initially everything worked fine (but I had asterisk 1.2 when I
>> installed the Openslug 3.10).
>> I don't remember when exactly I had this issue.
>>
>> Do you use the IP address or the FQDN of the asterisk for the
>> configured SIP proxy and registrar on the softphones?
>>
>> I was using the FQDN and I remember that I had to change it to the IP
>> address when I changed the ISP. Not sure but I suspect that this was
>> also the moment when I encountered your problem too.
>>
>>
>> Corneliu
>>
>> --- In nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>,
>> "Michael Drew" <blmdrew@...> wrote:
>> >
>> > Did you encounter the problem when you initially installed 3.10 with
>> 1.4.18?
>> >
>> > Just wondering if I should try that combination
>> >
>> > I did try the opt packages but same problem.
>> >
>> > I went back to unslung today with the optware/nslu2/cross/stable
>> feeds & is
>> > ok .
>> >
>> > On Tue, Oct 21, 2008 at 10:41 AM, Corneliu Doban
>> > <corneliu_doban@...>wrote:
>> >
>> > > I'm running openslug 3.10 with asterisk 1.4.18 and it works fine.
>> > > The problem sounds familiar but I don't remember what was the fix.
>> > >
>> > >
>> > > Corneliu
>> > >
>> > > --- In nslu2-asterisk@...<nslu2-asterisk%40yahoogroups.com>
>> <nslu2-asterisk%40yahoogroups.com>,
>> > > "Michael Drew" <blmdrew@> wrote:
>> > > >
>> > > > I Rebuilt with Ver 3.10 and Asterisk 4.16 and still same problem, so
>> > > as it
>> > > > stands:
>> > > > problem exists when I use - latest slugosbe and Asterisk V14.22
>> > > > OR: Ver3.10 with Asterisk
>> > > > V14.16
>> > > >
>> > > > It *does work* with Unslung (latest version) and Asterisk V14.22
>> > > >
>> > > > So I am not sure why there is a problem with OpenSlug and not with
>> > > > Unslung?????
>> > > > I can't see anything in the codec/conf files that would cause this.
>> > > >
>> > > > Just about ready to give in to Unslung
>> > > >
>> > > > Mike
>> > > >
>> > > >
>> > > > On Mon, Oct 20, 2008 at 8:11 PM, Michael Drew <blmdrew@> wrote:
>> > > >
>> > > > > I think that I will rebuild with Ver 3.10 to prove or disprove.
>> > > > > I did try again tonight with latest version. I thought that maybe
>> > > I was
>> > > > > missing something but still same problem
>> > > > > Will keep you posted.
>> > > > >
>> > > > > Thanks
>> > > > > Mike
>> > > > >
>> > > > > On Mon, Oct 20, 2008 at 8:02 PM, Michael Drew <blmdrew@> wrote:
>> > > > >
>> > > > >> I do have audio in the recordings i.e. my voice
>> > > > >>
>> > > > >>
>> > > > >> On Mon, Oct 20, 2008 at 5:05 PM, Corneliu Doban <corneliu_doban@
>> > > > >> > wrote:
>> > > > >>
>> > > > >>> I don't think that's something wrong with the slugosbe,
>> > > although I'm
>> > > > >>> running a pretty old version: 3.10
>> > > > >>>
>> > > > >>> You've said that is recording. Are those recordings blanks or
>> > > you you
>> > > > >>> have audio?
>> > > > >>>
>> > > > >>> Corneliu
>> > > > >>>
>> > > > >>>
>> > > > >>> --- In
>> nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com>
>> <nslu2-asterisk%40yahoogroups.com>
>> > > <nslu2-asterisk%40yahoogroups.com>,
>> > >
>> > > > >>> "Michael Drew" <blmdrew@> wrote:
>> > > > >>> >
>> > > > >>> > Thanks for the quick response.
>> > > > >>> > Yes, I had already tried this dtmfmode=rfc2833 to no avail.
>> > > > >>> > What I did end up doing was going to UnSlung and
>> everything worked
>> > > > >>> fine with
>> > > > >>> > same config.
>> > > > >>> >
>> > > > >>> > Than went back to slugosbe and same problem again???
>> > > > >>> >
>> > > > >>> > I would much rather go with Slugosbe but not sure why it works
>> > > with
>> > > > >>> Unslung
>> > > > >>> > & not with slugosbe.
>> > > > >>> >
>> > > > >>> >
>> > > > >>> > On Mon, Oct 20, 2008 at 11:04 AM, Corneliu Doban
>> > > > >>> > <corneliu_doban@>wrote:
>> > > > >>> >
>> > > > >>> > >
>> > > > >>> > > Try to add this to your phone configuration (in
>> sip.conf) to see
>> > > > >>> if helps:
>> > > > >>> > >
>> > > > >>> > > dtmfmode=rfc2833
>> > > > >>> > >
>> > > > >>> > > But this should be the default anyway (if no other mode is
>> > > set in the
>> > > > >>> > > [general] section).
>> > > > >>> > >
>> > > > >>> > > Good luck,
>> > > > >>> > > Corneliu
>> > > > >>> > >
>> > > > >>> > >
>> > > > >>> > > --- In
>> > > nslu2-asterisk@... <nslu2-asterisk%40yahoogroups.com><nslu2-asterisk%
>> 40yahoogroups.com>
>> > > <nslu2-asterisk%40yahoogroups.com>
>> > > > >>> <nslu2-asterisk%40yahoogroups.com>,
>> > > > >>>
>> > > > >>> > > "blmdrew" <blmdrew@> wrote:
>> > > > >>> > > >
>> > > > >>> > > > Hi.
>> > > > >>> > > > This is my first post on the forum.
>> > > > >>> > > > Have been home sick for the last few days and decided it
>> > > was a good
>> > > > >>> > > > time to try getting pbx on my Slug.I have a working PBX
>> > > > >>> (Asterisk 1.4)
>> > > > >>> > > > on another machine so I know that config is good.
>> > > > >>> > > >
>> > > > >>> > > > My problem is that my voicemail on the Slug is not
>> > > recognizing any
>> > > > >>> > > > incoming DTMFs from softphones. (It is recording messages
>> > > though)
>> > > > >>> > > >
>> > > > >>> > > > I know that the Softphones are set correctly as they work
>> > > on the
>> > > > >>> other
>> > > > >>> > > > PBX on the network.
>> > > > >>> > > >
>> > > > >>> > > > I am using Asterisk 1.4, SlugosBE with a 4 gb flash
>> > > drive(I know
>> > > > >>> that
>> > > > >>> > > > I don't need one that big)
>> > > > >>> > > >
>> > > > >>> > > > Here is my Sip.conf
>> > > > >>> > > >
>> > > > >>> > > > [general]
>> > > > >>> > > > port = 5060
>> > > > >>> > > > bindaddr = 0.0.0.0
>> > > > >>> > > > context = others
>> > > > >>> > > >
>> > > > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
>> > > > >>> > > > ;register => xxxxxx:xxxxxxx@:5060
>> > > > >>> > > >
>> > > > >>> > > > register => xxxxxx:xxxxxxx@:5060
>> > > > >>> > > > register => xxxxxx:xxxxxxx@:5060
>> > > > >>> > >
>> > > > >>> > > >
>> > > > >>> > > > [2000]
>> > > > >>> > > > type=friend
>> > > > >>> > > > context=my-phones
>> > > > >>> > > > secret=1234
>> > > > >>> > > > host=dynamic
>> > > > >>> > > >
>> > > > >>> > > > [2001]
>> > > > >>> > > > type=friend
>> > > > >>> > > > context=my-phones
>> > > > >>> > > > secret=1234
>> > > > >>> > > > host=dynamic
>> > > > >>> > > >
>> > > > >>> > > > [link2voip-sw1]
>> > > > >>> > > > context=from-voip-provider
>> > > > >>> > > > type=friend
>> > > > >>> > > > ;host=sip.ca1.link2voip.com
>> > > > >>> > > > host=sip.us1.link2voip.com
>> > > > >>> > > > username=xxxxxx
>> > > > >>> > > > secret=xxxxxxx
>> > > > >>> > > > canreinvite=no ; if using a nat, do not change
>> > > > >>> > > > insecure=port,invite ; do NOT remove this
>> > > > >>> > > > qualify=5000 ; do NOT remove this
>> > > > >>> > > > dtmfmode=auto
>> > > > >>> > > > nat=no ; do NOT remove/change this
>> > > > >>> > > > disallow=all
>> > > > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729
>> license or
>> > > > >>> can do
>> > > > >>> > > > passthru
>> > > > >>> > > > allow=ulaw
>> > > > >>> > > >
>> > > > >>> > > > [link2voip-sw2]
>> > > > >>> > > > context=from-voip-provider
>> > > > >>> > > > type=friend
>> > > > >>> > > > ;host=sip.ca2.link2voip.com
>> > > > >>> > > > host=sip.us2.link2voip.com
>> > > > >>> > > > username=xxxxxx
>> > > > >>> > > > secret=xxxxxxx
>> > > > >>> > > > canreinvite=no ; if using a nat, do not change
>> > > > >>> > > > insecure=port,invite ; do NOT remove this
>> > > > >>> > > > qualify=5000 ; do NOT remove this
>> > > > >>> > > > dtmfmode=auto
>> > > > >>> > > > nat=no ; do NOT remove/change this
>> > > > >>> > > > disallow=all
>> > > > >>> > > > ;allow=g729 ;uncomment if you have purchased a g729
>> license or
>> > > > >>> can do
>> > > > >>> > > > passthru
>> > > > >>> > > > allow=ulaw
>> > > > >>> > > >
>> > > > >>> > > > I have been scouring posts and googleing for 2 days to no
>> > > avail.
>> > > > >>> > > > Any help would be greatly appreciated
>> > > > >>> > > >
>> > > > >>> > > > Mike
>> > > > >>> > > >
>> > > > >>> > >
>> > > > >>> > >
>> > > > >>> > >
>> > > > >>> >
>> > > > >>>
>> > > > >>>
>> > > > >>>
>> > > > >>
>> > > > >>
>> > > > >
>> > > >
>> > >
>> > >
>> > >
>> >
>>
>>  
>>
>
>