[asterisk-bugs] [JIRA] (ASTERISK-23539) Crash when attempting to dial from a PJSIP endpoint
Dan Cropp (JIRA)
noreply at issues.asterisk.org
Wed Mar 26 10:13:18 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-23539?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Dan Cropp updated ASTERISK-23539:
---------------------------------
Attachment: backtrace.txt
This is the backtrace for the crash.
> Crash when attempting to dial from a PJSIP endpoint
> ---------------------------------------------------
>
> Key: ASTERISK-23539
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-23539
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip
> Affects Versions: 12.1.1
> Environment: Debian GNU/Linux 7.4 (wheezy)
> Reporter: Dan Cropp
> Severity: Critical
> Attachments: backtrace.txt
>
>
> 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:7001 at 192.168.9.142:5063' 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:7001 at 192.168.9.142:5063 ; 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()
> *CLI> pjsip set logger on
> PJSIP Logging enabled
> *CLI> sip set debug on
> SIP Debugging enabled
> *CLI> logger set level DEBUG on
> Logger status for 'DEBUG' has been set to 'on'.
> *CLI> <--- 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-6f6d1a0e
> From: "7001" <sip:7001 at 192.168.9.234>;tag=495a05232460c742o3
> To: "7001" <sip:7001 at 192.168.9.234>
> Call-ID: a93c73c5-83c75033 at 192.168.9.142
> CSeq: 27775 REGISTER
> Max-Forwards: 70
> Contact: "7001" <sip:7001 at 192.168.9.142:5063>;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-6f6d1a0e
> Call-ID: a93c73c5-83c75033 at 192.168.9.142
> From: "7001" <sip:7001 at 192.168.9.234>;tag=495a05232460c742o3
> To: "7001" <sip:7001 at 192.168.9.234>;tag=z9hG4bK-6f6d1a0e
> CSeq: 27775 REGISTER
> WWW-Authenticate: Digest realm="asterisk",nonce="1395843228/0688d35483a14f2d09d036995a88b2a3",opaque="0e34a1f012da413c",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-50261aae
> From: "7001" <sip:7001 at 192.168.9.234>;tag=495a05232460c742o3
> To: "7001" <sip:7001 at 192.168.9.234>
> Call-ID: a93c73c5-83c75033 at 192.168.9.142
> CSeq: 27776 REGISTER
> Max-Forwards: 70
> Authorization: Digest username="7001",realm="asterisk",nonce="1395843228/0688d35483a14f2d09d036995a88b2a3",uri="sip:192.168.9.234",algorithm=MD5,response="2ac24f75fd79956299bd3e2bb7f409d8",opaque="0e34a1f012da413c",qop=auth,nc=00000001,cnonce="c3cd3f56"
> Contact: "7001" <sip:7001 at 192.168.9.142:5063>;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:7001 at 192.168.9.142:5063' 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-50261aae
> Call-ID: a93c73c5-83c75033 at 192.168.9.142
> From: "7001" <sip:7001 at 192.168.9.234>;tag=495a05232460c742o3
> To: "7001" <sip:7001 at 192.168.9.234>;tag=z9hG4bK-50261aae
> CSeq: 27776 REGISTER
> Date: Wed, 26 Mar 2014 14:13:48 GMT
> Contact: <sip:7001 at 192.168.9.142:5063>;expires=3599
> Contact: <sip:7001 at 192.168.9.142:5063>
> Content-Length: 0
> <--- Received SIP request (900 bytes) from UDP:192.168.9.142:5063 --->
> INVITE sip:1 at 192.168.9.234 SIP/2.0
> Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-3dc3edde
> From: "7001" <sip:7001 at 192.168.9.234>;tag=bf73e9ea2ad3967fo3
> To: <sip:1 at 192.168.9.234>
> Call-ID: 31efa286-f45bd693 at 192.168.9.142
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: "7001" <sip:7001 at 192.168.9.142:5063>
> Expires: 240
> User-Agent: Cisco/SPA504G-7.4.8a
> Content-Length: 395
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
> Supported: replaces
> Content-Type: application/sdp
> v=0
> o=- 21730 21730 IN IP4 192.168.9.142
> s=-
> c=IN IP4 192.168.9.142
> t=0 0
> m=audio 16394 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-3dc3edde
> Call-ID: 31efa286-f45bd693 at 192.168.9.142
> From: "7001" <sip:7001 at 192.168.9.234>;tag=bf73e9ea2ad3967fo3
> To: <sip:1 at 192.168.9.234>;tag=z9hG4bK-3dc3edde
> CSeq: 101 INVITE
> WWW-Authenticate: Digest realm="asterisk",nonce="1395843406/8bcb262f8aca873f4c4be8edb2f19c46",opaque="33a814b90efd5a95",algorithm=md5,qop="auth"
> Content-Length: 0
> <--- Received SIP request (381 bytes) from UDP:192.168.9.142:5063 --->
> ACK sip:1 at 192.168.9.234 SIP/2.0
> Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-3dc3edde
> From: "7001" <sip:7001 at 192.168.9.234>;tag=bf73e9ea2ad3967fo3
> To: <sip:1 at 192.168.9.234>;tag=z9hG4bK-3dc3edde
> Call-ID: 31efa286-f45bd693 at 192.168.9.142
> CSeq: 101 ACK
> Max-Forwards: 70
> Contact: "7001" <sip:7001 at 192.168.9.142:5063>
> User-Agent: Cisco/SPA504G-7.4.8a
> Content-Length: 0
> <--- Received SIP request (1157 bytes) from UDP:192.168.9.142:5063 --->
> INVITE sip:1 at 192.168.9.234 SIP/2.0
> Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-45d986d2
> From: "7001" <sip:7001 at 192.168.9.234>;tag=bf73e9ea2ad3967fo3
> To: <sip:1 at 192.168.9.234>
> Call-ID: 31efa286-f45bd693 at 192.168.9.142
> CSeq: 102 INVITE
> Max-Forwards: 70
> Authorization: Digest username="7001",realm="asterisk",nonce="1395843406/8bcb262f8aca873f4c4be8edb2f19c46",uri="sip:1 at 192.168.9.234",algorithm=MD5,response="c26bdaf9161ab844fea4ac128745e8b5",opaque="33a814b90efd5a95",qop=auth,nc=00000001,cnonce="c318431f"
> Contact: "7001" <sip:7001 at 192.168.9.142:5063>
> Expires: 240
> User-Agent: Cisco/SPA504G-7.4.8a
> Content-Length: 395
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
> Supported: replaces
> Content-Type: application/sdp
> v=0
> o=- 21730 21730 IN IP4 192.168.9.142
> s=-
> c=IN IP4 192.168.9.142
> t=0 0
> m=audio 16394 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
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