On Sun, 1 Aug 2004 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=f7e5481bb929c765 > To: <sip:[EMAIL PROTECTED]>;tag=as269fa212 > Call-ID: [EMAIL PROTECTED] > CSeq: 28952 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > 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 [EMAIL PROTECTED] or whatever instead of using [EMAIL PROTECTED] (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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
