[A*UG] a2billing strangeness
rw at rickwhitley.com
Wed May 26 00:46:48 CDT 2010
Hi, I added the externip and localnet settings to my sip.conf file but I
am still getting the same behaviour. I'll look at it again tomorrow.
On Tue, 2010-05-25 at 21:25 -0500, Brett Nemeroff wrote:
> 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>
> 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
> 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
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> Supported: replaces.
> Contact: <sip:+12815492260 at 10.1.10.9>.
> Content-Type: application/sdp.
> Content-Length: 232.
> 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
> > 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
> > AAUG Web Site: http://aaug.bybent.com/
> "Worship is a journey with God as the goal."
"Worship is a journey with God as the goal."
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