Matthew Boehm wrote:
Yes, I have a exten => _X. in the context that the sip device has as context= in the sipusers table. However, the sipusers entry didn't get loaded and the device tried to use default.Well, for one, what you describe is NOT a bug in RealTime SIP. (damn I wish people would stop doing that.)Secondly, this does not belong on the developers list. Since your UA has obviously registered, SIP RT has done its job. Your problem now seems to be in your extensions. Do you have an extensions line for 18005551212? default just hangs up on all calls sent to it. So, IMHO, SIP RT hasn't done it's job when a device with entries in both the sippeers and sipusers families registers, and the sipusers family entry doesn't get loaded. As for an update. I restarted asterisk and everything seems to be working fine. I have yet to be able to reproduce this bug. However, I won't really be testing it until sunday, so I will update this thread sunday evening after we do extensive tests. -Chris -MatthewFrom: "Chris A. Icide" <[EMAIL PROTECTED]> Reply-To: Asterisk Developers Mailing List <[email protected]> Date: Fri, 12 Aug 2005 20:49:26 -0700 To: Asterisk Developers Mailing List <[email protected]> Subject: [Asterisk-Dev] Bug in realtime SIP Asterisk version: CVS Head from 8/11/05 extconfig.conf sipusers => mysql,zbx,sipusers sippeers => mysql,zbx,sippeers localhost*CLI> realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username zbx for 27 minutes, 20 seconds. realtime load sippeers name 104 and realtime load sipusers 104 both show the sip device information on screen. When the sip device registers (it's dynamic), sip show peers and sip show users show the following sip show peers 104/104 (Unspecified) D 255.255.255.255 0 Unmonitored sip show users No entry every shows up for users. If I try to make a call with the device, I get 403 Forbidden <-- SIP read from 192.168.254.16:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c From: Sipura2000-2 <sip:[EMAIL PROTECTED]>;tag=ffbcad7047bef41co1 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: Sipura2000-2 <sip:[EMAIL PROTECTED]:5060> Expires: 240 User-Agent: Sipura/SPA2000-2.0.10(c) Content-Length: 426 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 266317 266317 IN IP4 192.168.254.16 s=- c=IN IP4 192.168.254.16 t=0 0 m=audio 16384 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 19 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.254.16 : 5060 (non-NAT) Found no matching peer or user for '192.168.254.16:5060' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.16:16384 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 18005551212 in default list_route: hop: <sip:[EMAIL PROTECTED]:5060> Transmitting (no NAT) to 192.168.254.16:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c From: Sipura2000-2 <sip:[EMAIL PROTECTED]>;tag=ffbcad7047bef41co1 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- -- Executing Hangup("SIP/192.168.254.6-08a17090", "") in new stack == Spawn extension (default, 18005551212, 1) exited non-zero on 'SIP/192.168.254.6-08a17090' Reliably Transmitting (no NAT) to 192.168.254.16:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c From: Sipura2000-2 <sip:[EMAIL PROTECTED]>;tag=ffbcad7047bef41co1 To: <sip:[EMAIL PROTECTED]>;tag=as31d7c52b Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev_______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev |
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