[Asterisk-Users] Adding SIP Based Termination

[Greg Hill]

On Sun, 1 Aug 2004 sgup015 at ec.auckland.ac.nz wrote:

> 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.


403 forbidden usually means you didn't authenticate correctly to the other
SIP endpoint. IIRC, your sip.conf section for your this provider included
host, secret, and username. Sometimes you need to use fromuser and
fromdomain as well -- sometimes you're expected to identify yourself as
12345 at siptermination.com or whatever instead of using
asterisk at youriporhostname (this is what asterisk will use by default). You
would make asterisk identify itself the other way by using fromuser=12345
and fromdomain=siptermination.com in the appropriate section of your
sip.conf.

Give that a try and let us know what happens.. Another thing you could try
would be to make a softphone like x-ten lite, msn messenger, or one of the
linux varieties connect to your provider. Sometimes they're a little
easier to get working because they don't have so many little things you
can tweak. After you have a known good configuration there, you could do a
sip debug or network packet dump to see the communication it's making to
the provider, and then compare that with what asterisk says when it talks
to the provider.

Greg