Additional information with "pjsip set logger on" ------------------------- Register succeeds... -------------------------
<--- Received SIP request (485 bytes) from UDP:192.168.9.142:5063 ---> REGISTER sip:192.168.9.234 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-deea79e7 From: "7001" <sip:[email protected]>;tag=ee56a5177681851fo3 To: "7001" <sip:[email protected]> Call-ID: [email protected] CSeq: 25282 REGISTER Max-Forwards: 70 Contact: "7001" <sip:[email protected]:5063>;expires=3600 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces <--- Transmitting SIP response (469 bytes) to UDP:192.168.9.142:5063 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.9.142:5063;rport;received=192.168.9.142;branch=z9hG4bK-deea79e7 Call-ID: [email protected] From: "7001" <sip:[email protected]>;tag=ee56a5177681851fo3 To: "7001" <sip:[email protected]>;tag=z9hG4bK-deea79e7 CSeq: 25282 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1395782973/f72250272122471132aabf25deed1c0b",opaque="110098de72b0d893",algorithm=md5,qop="auth" Content-Length: 0 <--- Received SIP request (740 bytes) from UDP:192.168.9.142:5063 ---> REGISTER sip:192.168.9.234 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-f5a029e3 From: "7001" <sip:[email protected]>;tag=ee56a5177681851fo3 To: "7001" <sip:[email protected]> Call-ID: [email protected] CSeq: 25283 REGISTER Max-Forwards: 70 Authorization: Digest username="7001",realm="asterisk",nonce="1395782973/f72250272122471132aabf25deed1c0b",uri="sip:192.168.9.234",algorithm=MD5,response="e234a6e6abf82aec119d49a413e0a9b1",opaque="110098de72b0d893",qop=auth,nc=00000001,cnonce="9c4b3692" Contact: "7001" <sip:[email protected]:5063>;expires=3600 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces -- Added contact 'sip:[email protected]:5063' to AOR '7001' with expiration of 3600 seconds <--- Transmitting SIP response (442 bytes) to UDP:192.168.9.142:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.9.142:5063;rport;received=192.168.9.142;branch=z9hG4bK-f5a029e3 Call-ID: [email protected] From: "7001" <sip:[email protected]>;tag=ee56a5177681851fo3 To: "7001" <sip:[email protected]>;tag=z9hG4bK-f5a029e3 CSeq: 25283 REGISTER Date: Tue, 25 Mar 2014 21:29:33 GMT Contact: <sip:[email protected]:5063>;expires=3599 Contact: <sip:[email protected]:5063> Content-Length: 0 ------------------------- Dialing 1 from phone below. ------------------------- *CLI> <--- Received SIP request (898 bytes) from UDP:192.168.9.142:5063 ---> INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-9b8d1e07 From: "7001" <sip:[email protected]>;tag=9fa6d06bfc4546d4o3 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 101 INVITE Max-Forwards: 70 Contact: "7001" <sip:[email protected]:5063> Expires: 240 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 8644 8644 IN IP4 192.168.9.142 s=- c=IN IP4 192.168.9.142 t=0 0 m=audio 16462 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 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:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <--- Transmitting SIP response (455 bytes) to UDP:192.168.9.142:5063 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.9.142:5063;rport;received=192.168.9.142;branch=z9hG4bK-9b8d1e07 Call-ID: [email protected] From: "7001" <sip:[email protected]>;tag=9fa6d06bfc4546d4o3 To: <sip:[email protected]>;tag=z9hG4bK-9b8d1e07 CSeq: 101 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1395783027/7ae9aaf5d61fc322eac8dec60d9c8dbe",opaque="13d5988e59a920a6",algorithm=md5,qop="auth" Content-Length: 0 <--- Received SIP request (381 bytes) from UDP:192.168.9.142:5063 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-9b8d1e07 From: "7001" <sip:[email protected]>;tag=9fa6d06bfc4546d4o3 To: <sip:[email protected]>;tag=z9hG4bK-9b8d1e07 Call-ID: [email protected] CSeq: 101 ACK Max-Forwards: 70 Contact: "7001" <sip:[email protected]:5063> User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 0 <--- Received SIP request (1155 bytes) from UDP:192.168.9.142:5063 ---> INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-d1aac763 From: "7001" <sip:[email protected]>;tag=9fa6d06bfc4546d4o3 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="7001",realm="asterisk",nonce="1395783027/7ae9aaf5d61fc322eac8dec60d9c8dbe",uri="sip:[email protected]",algorithm=MD5,response="c0f7e47e6af69559a266c3ec22793ff0",opaque="13d5988e59a920a6",qop=auth,nc=00000001,cnonce="9adbf5ea" Contact: "7001" <sip:[email protected]:5063> Expires: 240 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 8644 8644 IN IP4 192.168.9.142 s=- c=IN IP4 192.168.9.142 t=0 0 m=audio 16462 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 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:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------- Asterisk 12.1.1 Crashes at this point ------------------------- From: [email protected] [mailto:[email protected]] On Behalf Of Dan Cropp Sent: Tuesday, March 25, 2014 4:22 PM To: [email protected] Subject: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:[email protected]:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to look or how to track down the problem. Can anyone offer some hints? --------------------- pjsip.conf --------------------- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [7001] type=endpoint transport=transport-udp context=IS disallow=all allow=ulaw auth=7001 aors=7001 [7001] type=aor max_contacts=1 contact=sip:[email protected]:5063 ; Line 4 on my phone is setup for port 5063. ; I have also tried without this setting and am seeing the exact same scenario [7001] type=auth auth_type=userpass password=1234 username=7001 --------------------- extensions.conf --------------------- [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=DAHDI/G2 ; Trunk interface TRUNKMSD=1 [IS] exten => 1,1,Verbose(1,Unrouted call handler) exten => 1,n,Answer() exten => 1,n,Wait(1) exten => 1,n,Playback(tt-weasels) exten => 1,n,Hangup() Have a great day! Dan
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