Ajit wrote:
Hi, I am trying to use the manager API to originate a SIP call from one asterisk extension to another. eg extension [EMAIL PROTECTED] calls extension [EMAIL PROTECTED] this fails with a 482 "Loop detected".I beleive this behaviour is incorrect according to rfc3261 : The UAS processes the first such request received and responds with a 482 (Loop Detected) to the rest of them. In this case its a sent request colliding with the first recieved request. Can people on the dev list comment on if this is a bug and where in the code it may be fixed. Note that i do not need SIP proxy functionality or NAT from asterix. Any ideas on workarounds?? I think if i can have two SIP stacks on different ports calling each other this behaviour can be avoided.Is this possible and if so how can i configure this. Regards, Ajit P.S: Here is the SIP debug (ip addresses modified) LI> sip debug SIP Debugging enabled == Parsing '/etc/asterisk/manager.conf': Found == Manager 'manager' logged on from xxx.yyy.xxx.yyy We're at myasteriskserver port 10878 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to myasteriskserver:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 28 Apr 2006 13:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 5351 5351 IN IP4 myasteriskserver s=session c=IN IP4 myasteriskserver t=0 0 m=audio 10878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from myasteriskserver:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 28 Apr 2006 13:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 5351 5351 IN IP4 myasteriskserver s=session c=IN IP4 myasteriskserver t=0 0 m=audio 10878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (13 headers 10 lines)--- Transmitting (no NAT) to myasteriskserver:5060: SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport;received=myasteriskserver From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 To: <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- <-- SIP read from myasteriskserver:5060: SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport;received=myasteriskserver From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 To: <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- (10 headers 0 lines)--- -- Got SIP response 482 "Loop Detected" back from myasteriskserver Transmitting (no NAT) to myasteriskserver:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 To: <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <-- SIP read from myasteriskserver:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP myasteriskserver:5060;branch=z9hG4bK7aa5b7b4;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 To: <sip:[EMAIL PROTECTED]>;tag=as7c2a32b7 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' == Manager 'manager' logged off from xxx.yyy.xxx.yyy _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Why don't you use something like the chan_local channel driver to send the call into the dialplan where it will then execute the extension? If you don't you're going to see what you're getting above. You're looping an outbound call back inbound to the same box, with the same callid... so it's perceiving it as a loop.
-- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
