I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office.
I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to route certain calls to Asterisk. However, the calls are placed in the "from-sip-external" context that Asterisk @home uses for unidentified SIP calls and are subsequently dropped. I can make the calls connect by modifying the from-sip-external context, but I would like to be able to specify that calls from the router (on a static IP) are placed in a different context. Here is part of my sip_additional.conf: [Cisco1760_mc] type=friend host=192.168.0.254 context=from-pstn disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes qualify=yes However, when I use sip debug to monitor an attempted call (711515 from one of the phones connected to the Cisco), these lines appears in part of the debug: Found no matching peer or user for '192.168.0.254:53464' Looking for 711515 in from-sip-external Shouldn't it match Cisco1760_mc? I've included the full debug below. Thanks in advance for your help. I'm happy to provide any additional information. --TWH Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: "Tim Howell" <sip:[EMAIL PROTECTED]>;tag=49582394-1813 To: <sip:[EMAIL PROTECTED]> Date: Tue, 15 Mar 2005 17:12:08 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 838919162-2494304729-2389872912-2937628716 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY , INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: "Tim Howell" <sip:[EMAIL PROTECTED]>;party=calling;screen=no;pr ivacy=off Timestamp: 1110906728 Contact: <sip:[EMAIL PROTECTED]:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 267 v=0 o=CiscoSystemsSIP-GW-UserAgent 6054 4992 IN IP4 192.168.0.254 s=SIP Call c=IN IP4 192.168.0.254 t=0 0 m=audio 16946 RTP/AVP 0 100 19 c=IN IP4 192.168.0.254 a=rtpmap:0 PCMU/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:19 CN/8000 a=ptime:20 20 headers, 12 lines Using latest request as basis request Sending to 192.168.0.254 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 19 Peer audio RTP is at port 192.168.0.254:16946 Found description format PCMU Found description format X-NSE Found description format CN Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing) , combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x2 (gsm), combined - 0x0 (noth ing) Found no matching peer or user for '192.168.0.254:53464' Looking for 711515 in from-sip-external list_route: hop: <sip:[EMAIL PROTECTED]:5060> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: "Tim Howell" <sip:[EMAIL PROTECTED]>;tag=49582394-1813 To: <sip:[EMAIL PROTECTED]>;tag=as2a29cbba Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.254:5060 -- Executing AbsoluteTimeout("SIP/192.168.0.254-094a8648", "15") in new sta ck -- Set Absolute Timeout to 15 -- Executing Congestion("SIP/192.168.0.254-094a8648", "") in new stack Transmitting (no NAT): SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: "Tim Howell" <sip:[EMAIL PROTECTED]>;tag=49582394-1813 To: <sip:[EMAIL PROTECTED]>;tag=as2a29cbba Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.254:5060 == Spawn extension (from-sip-external, 711515, 2) exited non-zero on 'SIP/192 .168.0.254-094a8648' -- Executing AbsoluteTimeout("SIP/192.168.0.254-094a8648", "15") in new sta ck -- Set Absolute Timeout to 15 -- Executing Congestion("SIP/192.168.0.254-094a8648", "") in new stack == Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/192.168. 0.254-094a8648' asterisk1*CLI> Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: "Tim Howell" <sip:[EMAIL PROTECTED]>;tag=49582394-1813 To: <sip:[EMAIL PROTECTED]>;tag=as2a29cbba Date: Tue, 15 Mar 2005 17:12:08 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 9 headers, 0 lines Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> sip no debug SIP Debugging Disabled asterisk1*CLI> _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users