[Asterisk-Users] Adding SIP Based Termination

[sgup015 at ec.auckland.ac.nz]

Hi,
I've had a look at it and the timeout error is what happens straight after the
phone disconnects:

Aug  1 04:07:13 WARNING[106511]: pbx.c:922 pbx_substitute_variables_temp: The
use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo'
Aug  1 04:07:20 WARNING[5126]: chan_sip.c:673 retrans_pkt: Maximum retries
exceeded on call 53bc2ead1d050bbe003bead11adaaa25 at 202.36.23.70 for seqno 102
(Non-critical Request)
Aug  1 04:07:23 WARNING[106511]: pbx.c:1924 ast_pbx_run: Timeout, but no rule
't' in context 'sip'


Aug  1 04:10:01 WARNING[109583]: pbx.c:1924 ast_pbx_run: Timeout, but no rule
't' in context 'sip'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 219.88.229.122;branch=z9hG4bKb552afc0d1c81034
From: "Sahil Gupta" <sip:sahil-akl at 202.36.23.70>;tag=f7e5481bb929c765
To: <sip:346478340015 at 202.36.23.70>;tag=as269fa212
Call-ID: cbfa1d49713e0039 at 219.88.229.122
CSeq: 28952 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:346478340015 at 202.36.23.70>
Content-Length: 0

to 219.88.229.122:5060

Any ideas on that error?  A quick search on google didn't bring up much.

Cheers,
Sahil
Quoting Greg Hill <gregh-asterisk at hillnet.us>:

> On Sun, 1 Aug 2004 sgup015 at ec.auckland.ac.nz wrote:
>
> > exten => _34.,1,Dial(SIP/${EXTEN-1}@prov,20,tr)
>
> The r at the end of this line tells asterisk to generate a ringing sound
> for you to hear. In other words, the ringing you're hearing isn't coming
> from the far end SIP device. Taking the r out will probably help you get a
> little closer to the solution.. another thing you can do is turn sip debug
> on while you try to place the call and see what happens. Watch for
> responses from the far end sip server, etc.
>
> Greg
>
>
>
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