Carlos Chavez
2012-Feb-09 18:50 UTC
[asterisk-users] Problem with SIP phone outside local network
I am having a strange problem with an external SIP phone. It can register and receive calls but it cannot initiate any calls. A softphone on the same network works without problems. As far as I can notice the difference is that the hard phone is not sending the proper contact info. In the fullcontact field I can see its private IP address "sip:1008 at 192.168.2.18:5060^3Btransport=udp" while the softphone provides the public IP. The hard phone is an Aastra 6730i. A similar phone can make and receive calls when connected from another external network so I do not think it is an Aastra issue. Asterisk is behind a NAT (on DMZ) and has the proper externhost. The sip phone definition has nat=yes Any ideas? Here is a sip debug of the failed call: <--- SIP read from UDP:201.141.67.189:41528 ---> INVITE sip:1007 at pbxwbu.xxxxx.org:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6 Max-Forwards: 70 From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone> Call-ID: 9567bd1f0b345f53 CSeq: 29187 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "1008" <sip:1008 at 192.168.2.18:5060;transport=udp>; +sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D21B027>" Supported: path, 100rel, replaces User-Agent: Aastra 6730i/3.2.2.1136 Content-Type: application/sdp Content-Length: 595 v=0 o=MxSIP 0 1 IN IP4 192.168.2.18 s=SIP Call c=IN IP4 192.168.2.18 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (14 headers 25 lines) --- Sending to 201.141.67.189:41528 (NAT) Using INVITE request as basis request - 9567bd1f0b345f53 Found peer '1008' for '1008' from 201.141.67.189:41528 <--- Reliably Transmitting (NAT) to 201.141.67.189:41528 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6;received=201.141.67.189;rport=41528 From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 INVITE Server: Asterisk PBX 1.8.9.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="553cdc38" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9567bd1f0b345f53' in 7040 ms (Method: INVITE) Retransmitting #1 (NAT) to 201.141.67.189:41528: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6;received=201.141.67.189;rport=41528 From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 INVITE Server: Asterisk PBX 1.8.9.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="553cdc38" Content-Length: 0 --- <--- SIP read from UDP:201.141.67.189:41528 ---> ACK sip:1007 at pbxwbu.xxxxx.org:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6 Max-Forwards: 70 From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 ACK User-Agent: Aastra 6730i/3.2.2.1136 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:201.141.67.189:41528 ---> ACK sip:1007 at pbxwbu.xxxxx.org:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6 Max-Forwards: 70 From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 ACK User-Agent: Aastra 6730i/3.2.2.1136 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- == Using SIP RTP CoS mark 5 -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120209/de6f06a6/attachment.pgp>
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