Brenton, Yves, ... I've located the cause of the problem in chan_sip.c but am still trying to find the exact cause being completely new to the asterisk code. It seems that there was an added function in 1.135 called 'find_user' that is supposed to lookup the users incoming call limit but the routine is unable to find a matching user for my AS5300 which I suspect is because it does not REGISTER with the server prior to attempting to send calls.
I'm going to continue debugging a little later and see if I can narrow it down more ... Adam -----Original Message----- From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: 30/07/03 14:09 Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 Hi, I am using the latest cvs release of asterisk, and the behaviour is in fact the same, outbound calls work fine, but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked" by asterisk, and never reach the phone. The setup is the same : 7960 <------> asterisk <------> C2651<-----> PSTN Yves |---------+-------------------------------------> | | "Low, Adam" | | | <[EMAIL PROTECTED]>| | | Sent by: | | | [EMAIL PROTECTED]| | | .digium.com | | | | | | | | | 30/07/2003 11:37 | | | Please respond to | | | asterisk-users | | | | |---------+-------------------------------------> >----------------------------------------------------------------------- ------------------------------------------------| | | | To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> | | cc: | | Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 | >----------------------------------------------------------------------- ------------------------------------------------| All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: "611012210" <sip:[EMAIL PROTECTED]> To: <sip:[EMAIL PROTECTED];user=phone;phone-context=unknown> Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: <sip:[EMAIL PROTECTED]:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person _______________________________________________ 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 ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
