Hi, I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. so Asterisk server 1 (192.168.250.111) sip.conf contains: [interboxsip] type=peer host=192.168.250.112 context=mycontext Asterisk server 2 (192.168.250.112) sip.conf contains: [interboxsip] type=peer host=192.168.250.111 context=mycontext I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666 in server 1 (192.168.250.111) via the interboxsip SIP trunk. The call fails and according to the SIP messages it seems to be an authentication problem. What am I missing? SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call): -- Executing [3...@from-internal:2] Dial("SIP/4053-00006dea", "SIP/interboxsip/3666|300|rt") in new stack Audio is at 192.168.250.112 port 15850 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.250.111:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport From: "device" <sip:[email protected]>;tag=as4d17a185 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:13:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 15850 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called interboxsip/3666 <--- SIP read from 192.168.250.111:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060 From: "device" <sip:[email protected]>;tag=as4d17a185 To: <sip:[email protected]>;tag=as00842b82 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2545a5dd" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 192.168.250.111:5060: ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport From: "device" <sip:[email protected]>;tag=as4d17a185 To: <sip:[email protected]>;tag=as00842b82 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/interboxsip-00006deb is circuit-busy SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): <-- SIP read from 192.168.250.112:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: "device" <sip:[email protected]>;tag=as18a568d6 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [email protected] Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: "device" <sip:[email protected]>;tag=as18a568d6 To: <sip:[email protected]>;tag=as57a19dac Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1327c5b6" Content-Length: 0 --- Scheduling destruction of call '[email protected]' in 15000 ms Found user '4053' <-- SIP read from 192.168.250.112:5060: ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: "device" <sip:[email protected]>;tag=as18a568d6 To: <sip:[email protected]>;tag=as57a19dac Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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