[A*UG] a2billing strangeness

[Brett Nemeroff]

You'll need to show the SDP in your invite.

If what your provider says is indeed true, then I'd expect you are getting
one way audio.

If you are, and your asterisk box is behind NAT (in other words, you have a
private IP not assigned by your voip provider) then you likely are missing a
externip declaration in sip.conf


On Tue, May 25, 2010 at 9:20 PM, rick whitley <rw at rickwhitley.com> wrote:

> I spoke with the sip provider and they tell me i am requesting rtp to my
> private ip number and not my public one. i added nat=yes to my sip
> config but i still have the same problem. does anyone know where to set
> the route to request rtp? here is a portion of the log file from the
> provider.
>
> .
> .
> .
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:+12815492260 at 10.1.10.9 <sip%3A%2B12815492260 at 10.1.10.9>>.
> Content-Type: application/sdp.
> Content-Length: 232.
> .
> .
> .
>
> thanks
>
>
>
>
> On Mon, 2010-05-24 at 15:48 -0500, Brett Nemeroff wrote:
> > Rick,
> > This is definitely a dtmfmode issue as Gil pointed out.
> >
> >
> > Try using "auto" for a DTMF mode and see if that helps. If not, there
> > are a number of "known" incompatibilities with certain vendors (Sonus
> > for example) with G711 and RFC2833.
> >
> >
> > For me, the most reliable has been:
> >
> >
> > Codec=G711 / DTMFmode=inband
> > Codec=G729 / DTMFmode=RFC2833
> > Codec=mixed / DTMFmode=auto
> >
> >
> > Good luck. Let us know what works for you.
> > -Brett
> >
> >
> >
> > On Mon, May 24, 2010 at 7:30 AM, rick whitley <rw at rickwhitley.com>
> > wrote:
> >         Hi, my name is Rick Whitley and I am new to the list.
> >
> >         I am running elastix 1.5.2-2 with a2billing 1.3.0. I have 2
> >         dahdi trunks
> >         and 2 sip trunks. when i call into one of the dahdi trunks, it
> >         is passed
> >         to a2billing and i get the "enter pin" prompt. it all works as
> >         it
> >         should. when i call one of the sip trunks, it is passed to
> >         a2billing, i
> >         get the "enter pin" prompt but a2billing doesn't seem to know
> >         that i am
> >         entering a number. before i finish entering the pin i get a
> >         "no pin
> >         entered" and the code stops. any thoughts as to why this would
> >         happen?
> >         --
> >         "Worship is a journey with God as the goal."
> >         rw
> >         rom.5:8
> >         www.rickwhitley.com
> >         www.daysjourney.org
> >
> >         _______________________________________________
> >         Austin-Asterisk-Users-Group mailing list
> >         Austin-Asterisk-Users-Group at bybent.com
> >         http://bybent.com/mailman/listinfo/austin-asterisk-users-group
> >         AAUG Web Site: http://aaug.bybent.com/
> >
> >
>
> --
> "Worship is a journey with God as the goal."
> rw
> rom.5:8
> www.rickwhitley.com
> www.daysjourney.org
>
>
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