Hi, When I receive an incoming call on a VoIPtalk account using ReceiveCall, I get a barrage of messages like this in the log:
"2008-09-19T13:23:17.479000Z":394:KERNEL:WARNING:vpc-vs4sipxtapi::00000000:sipXtapi:"OsMsgPool::FindFreeMsg 'MediaSignals' queue size (33) exceeds soft limit (32)" These continue to "2008-09-19T13:23:24.482000Z":432:KERNEL:WARNING:vpc-vs4sipxtapi::00000000:sipXtapi:"OsMsgPool::FindFreeMsg 'MediaSignals' queue size (64) exceeds soft limit (32)" and finally "2008-09-19T13:23:24.482000Z":433:KERNEL:CRIT:vpc-vs4sipxtapi::00000000:sipXtapi:"OsMsgPool::FindFreeMsg 'MediaSignals' queue size (64) exceeds hard limit (64)" In addition the CPU usage is pegged at 100% until I kill the process, preventing call operation (and most other computer usage). The problem does not occur when ReceiveCall is registered with a SipPhone account. The problem also occurs in my application. I found a couple of hits via Google on the error message. One was thought to be tied to a specific small platform having insufficient CPU resources (although this was not proven, I think). Another was a reported bug that was closed much later because a tester couldn't duplicate it (http://track.sipfoundry.org/browse/XPB-491). The fact that the problem occurs when registered with one SIP account but not another suggests (to me) that the problem is not due to insufficient CPU power or message queue entries. In addition, I sometimes get an access violation here: // Returns the flow graph that contains this resource or NULL if the // resource is not presently part of any flow graph. MpFlowGraphBase* MpResource::getFlowGraph(void) const { return mpFlowGraph; // <--access violation here } Platform: Windows XP Machine: VMware VM, 512 MB RAM, running on Dell T3400 quad-core 4GB RAM Branch: http://scm.sipfoundry.org/rep/sipX/branches/sipXtapi Source: current as of morning of 2008-09-19 Here is the INVITE when registered with VoIPtalk: ----Remote Host:217.14.138.177---- Port: 5065---- INVITE sip:[EMAIL PROTECTED];LINEID=e05d9e2471f3 SIP/2.0 To: <sip:[EMAIL PROTECTED]> From: <sip:[EMAIL PROTECTED]>;tag=04791737 Via: SIP/2.0/UDP 217.14.138.177:5065;branch=z9hG4bK-d87543-73f165122385fc75bb56-1-cHAzYWE4Njc0OGEzYjJmNDRhNjZmMA..-d87543- Call-ID: 5c42ac9b35bf8670a6d15c635c2f6757 CSeq: 371449624 INVITE Contact: <sip:[EMAIL PROTECTED]:5065> Max-Forwards: 67 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp User-Agent: WinGizmo/2.0.02 (Gizmo-s2n1) Content-Length: 426 JabberID: [EMAIL PROTECTED] CQBM: 242 RemoteIP: 198.65.166.156 P-hint: outbound, call to user v=0 o=GizmoProject 205289829 2059736434 IN IP4 217.14.138.177 s=GizmoAudioSession c=IN IP4 217.14.138.177 t=0 0 m=audio 16890 RTP/AVP 103 102 0 8 3 106 13 a=rtcp:6257 a=rtpmap:103 ISAC/16000 a=rtpmap:102 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:106 telephone-event/8000 a=rtpmap:13 CN/8000 m=video 28352 RTP/AVP 34 125 a=rtpmap:34 H263/90000 a=rtpmap:125 MP4V-ES/90000 ====================END==================== Here is the INVITE when registered with SipPhone: ----Remote Host:198.65.166.131---- Port: 5060---- INVITE sip:[EMAIL PROTECTED]:24568;LINEID=5fb7afa459b6 SIP/2.0 Record-Route: <sip:198.65.166.131;lr;ftag=19247c11> Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKb98c.e39c0d7.0 Via: SIP/2.0/UDP 198.65.166.156:37285 Via: SIP/2.0/UDP 192.168.0.11:64064;branch=z9hG4bK-d87543-e04a8251394ea234-1--d87543-;rport Max-Forwards: 69 Contact: <sip:[EMAIL PROTECTED]:37285> To: <sip:[EMAIL PROTECTED]> From: <sip:[EMAIL PROTECTED]>;tag=19247c11 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp User-Agent: WinGizmo/2.0.02 (Gizmo-s2n1) Content-Length: 418 JabberID: [EMAIL PROTECTED] CQBM: 242 RemoteIP: 198.65.166.156 v=0 o=GizmoProject 1817190632 1 IN IP4 198.104.137.136 s=GizmoAudioSession c=IN IP4 198.104.137.136 t=0 0 m=audio 6870 RTP/AVP 103 102 0 8 3 106 13 a=rtpmap:103 ISAC/16000 a=rtpmap:102 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:106 telephone-event/8000 a=rtpmap:13 CN/8000 a=rtcp:6871 m=video 6870 RTP/AVP 34 125 a=rtpmap:34 H263/90000 a=rtpmap:125 MP4V-ES/90000 ====================END==================== The logs are too big to attach, but they are available at: < http://www.ascendis.com/temp/ReceiveCall%202008-09-19%20%209-43-13.log > < http://www.ascendis.com/temp/ReceiveCall%202008-09-19%20%209-47-00.log > Does anyone have any ideas? Finest regards, Bill Root _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
