I experienced the exact same symptoms but didn't have the confidence to post my experience to this list because of my lack of experience with Asterisk. I restored the June 1 version from CVS and the problem went away. There's definitely a problem in code since June 1.
On Sat, 7 Jun 2003, John Todd wrote: > - After that point, all other SIP calls from any other device fail, > and looking at tethereal I see that there are no replies to new SIP > REGISTER requests, either. I can type "stop now" or "stop > gracefully" and the system will not stop. I have to manually killall > to get asterisk to die. > > - I backed out to a version from June 3 21:18 and all dial modes work > correctly with exactly the same /etc/asterisk/* files, so it is a > change in Asterisk and not in the phones. > > > > > > *CLI> show version > Asterisk CVS-06/07/03-01:40:15 built by [EMAIL PROTECTED] on a > i686 running Linux > *CLI> > *CLI> sip debug > SIP Debugging Enabled > Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 10.0.1.25:5060 > From: <sip:[EMAIL PROTECTED];user=phone>;tag=2961659159 > To: <sip:[EMAIL PROTECTED];user=phone> > Call-ID: [EMAIL PROTECTED] > CSeq: 1 INVITE > Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> > User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) > Expires: 300 > Content-Length: 243 > Content-Type: application/sdp > > v=0 > o=2204 23257 23257 IN IP4 10.0.1.25 > s=ATA186 Call > c=IN IP4 10.0.1.25 > t=0 0 > m=audio 16386 RTP/AVP 18 8 0 101 > a=rtpmap:18 G729/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 11 headers, 11 lines > Using latest request as basis request > Sending to 10.0.1.25 : 5060 (non-NAT) > Capabilities: us - 14, them - 268, combined - 12 > Non-codec capabilities: us - 1, them - 1, combined - 1 > > *CLI> > *CLI> > *CLI> show channels > Channel (Context Extension Pri ) State Appl. > Data > 0 active channel(s) > *CLI> > *CLI> > *CLI> sip show channels > Peer Username Call ID Seq (Tx/Rx) Lag Jitter Format > 10.0.1.25 (None) 1852710522@ 00101/00002 00000ms 0000ms 0 > 1 active SIP channel(s) > *CLI> > > > > Configuration for ATA-186 line 1: > > [2204] > type=friend > username=2204 > secret=somepassword > mailbox=2203 > host=dynamic > context=intern > canreinvite=no > dtmfmode=rfc2833 > nat=1 > > > > For reference, here is the SIP debug for a functional call from a > 7960 on the same version of Asterisk code (2203 = 7960, 2204 = > ATA-186 line 1) > > *CLI> > Sip read: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 10.0.1.15:5060 > From: "2203" <sip:[EMAIL PROTECTED]>;tag=0002b9eb0ef400c3289c4132-36211630 > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > Date: Sat, 07 Jun 2003 19:19:33 GMT > CSeq: 101 INVITE > User-Agent: CSCO/4 > Contact: <sip:[EMAIL PROTECTED]:5060> > Expires: 180 > Content-Type: application/sdp > Content-Length: 241 > Accept: application/sdp > Remote-Party-ID: "2203" > <sip:[EMAIL PROTECTED]>;party=calling;id-type=subscriber;privacy=off;screen=no > > v=0 > o=Cisco-SIPUA 14891 19200 IN IP4 10.0.1.15 > s=SIP Call > c=IN IP4 10.0.1.15 > t=0 0 > m=audio 23764 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 14 headers, 11 lines > Using latest request as basis request > Sending to 10.0.1.15 : 5060 (non-NAT) > Capabilities: us - 14, them - 268, combined - 12 > Non-codec capabilities: us - 1, them - 1, combined - 1 > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
