[A*UG] a2billing strangeness
rw at rickwhitley.com
Tue May 25 22:20:55 CDT 2010
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
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:+12815492260 at 10.1.10.9>.
On Mon, 2010-05-24 at 15:48 -0500, Brett Nemeroff wrote:
> 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.
> On Mon, May 24, 2010 at 7:30 AM, rick whitley <rw at rickwhitley.com>
> 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
> 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
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