[Asterisk-Users] SIP immediate hangups with latest CVS

Sat Jul 12 10:51:53 MST 2003

No change.  I am unable to use SIP at all, apparently, in this latest revision.

JT

>I had this a while back, and set canreinvite=no, and it fixed it.
>
>-d
>
>At 08:42 PM 7/11/2003 -0700, you wrote:
>
>>I've been banging my head on this for several hours, and I have no 
>>idea what's going on.   Maybe there is a very simple result, and 
>>I've been looking too hard at this this evening.  This is a brand 
>>new system, and I'm wondering if there have been SIP bugs 
>>introduced in the latest CVS that are preventing from working what 
>>should be a stupendously simple test.
>>
>>- Cisco 7960 (non-NATed)
>>- RH 8.0
>>- Asterisk CVS update as of ~8:00 PM EDT
>>- full "make clean; make install" on [asterisk,zaptel,libpri]
>>- 2ghz box with E1 card (that's pretty much not part of the equation)
>>
>>I have boiled the configuration down to an extremely (_extremely_) 
>>simple setup, and it does not work.  SIP calls from the 7960 are 
>>hanging up almost immediately, with no audio getting through.   It 
>>seems that the hangup happens just after the moment that the 7960 
>>sends the ACK message (judging from the debug below, at least.)  I 
>>have verified that demo-congrats is there, as my original problem 
>>stemmed from strange behavior with Zap dialing, and I kept 
>>simplifying, so this is the culmination of winnowing down the 
>>options to the most basic config.  The same phone works flawlessly 
>>with other lines that are configured on it to other * servers.
>>
>>Here is my entire relevant configuration.  It's as simple as you 
>>can get, really.  I dial 14109850123 (as a test number - it matches 
>>the _1X. list) and I get an almost instant hangup.
>>
>>---------------
>>;sip.conf
>>[general]
>>port = 5060                     ; Port to bind to
>>bindaddr = 0.0.0.0              ; Address to bind to
>>context = default               ; Default for incoming calls
>>dtmfmode=rfc2833
>>allow=all
>>
>>[3015321510]
>>type=friend
>>username=3015321510
>>secret=fluffernutter
>>host=dynamic
>>context=from-sip
>>allow=all
>>---------------
>>;extensions.conf
>>
>>[general]
>>static=yes
>>writeprotect=yes
>>
>>[from-sip]
>>exten => _1X.,1,SetCallerID(3015321510)
>>exten => _1X.,2,Answer
>>exten => _1X.,3,Playback(demo-congrats)
>>exten => h,1,Hangup
>>exten => t,1,Hangup
>>exten => i,1,Hangup
>>---------------
>>
>>Other strange notes:
>>  - quite often, when launching with "-vvvvgcd" I get a segfault.  I 
>>have the cores, if anyone is interested.
>>  - I have almost identical systems (same hardware, same MB, etc.) 
>>churning away with no problems with slightly older revs of code
>>
>>
>>
>>*CLI>
>>Sip read:
>>INVITE sip:14109850123 at 64.33.1.8 SIP/2.0
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>Date: Sat, 12 Jul 2003 03:24:34 GMT
>>CSeq: 101 INVITE
>>User-Agent: CSCO/4
>>Contact: <sip:3015321510 at 128.151.224.33:5060>
>>Expires: 180
>>Content-Type: application/sdp
>>Content-Length: 247
>>Accept: application/sdp
>>Remote-Party-ID: "3015321510" 
>><sip:3015321510 at 128.151.224.33>;party=calling;id-type=subscriber;privacy=off;screen=no
>>
>>v=0
>>o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33
>>s=SIP Call
>>c=IN IP4 128.151.224.33
>>t=0 0
>>m=audio 19364 RTP/AVP 0 8 18 101
>>a=rtpmap:0 PCMU/8000
>>a=rtpmap:8 PCMA/8000
>>a=rtpmap:18 G729/8000
>>a=rtpmap:101 telephone-event/8000
>>a=fmtp:101 0-15
>>
>>14 headers, 11 lines
>>Using latest request as basis request
>>Sending to 128.151.224.33 : 5060 (non-NAT)
>>Found audio format 0
>>Found audio format 8
>>Found audio format 18
>>Found audio format 101
>>Found description format PCMU
>>Found description format PCMA
>>Found description format G729
>>Found description format telephone-event
>>Capabilities: us - 2147483647, them - 268/0, combined - 268
>>Non-codec capabilities: us - 1, them - 1, combined - 1
>>Reliably Transmitting (no NAT):
>>SIP/2.0 407 Proxy Authentication Required
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>;tag=as74174b76
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>CSeq: 101 INVITE
>>User-Agent: Asterisk PBX
>>Contact:
>>Proxy-Authenticate: Digest realm="asterisk", nonce="2c9c06be"
>>Content-Length: 0
>>
>>
>>  to 128.151.224.33:5060
>>Sip read:
>>ACK sip:14109850123 at 64.33.1.8 SIP/2.0
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>;tag=as74174b76
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>Date: Sat, 12 Jul 2003 03:24:34 GMT
>>CSeq: 101 ACK
>>Content-Length: 0
>>
>>
>>8 headers, 0 lines
>>Sip read:
>>INVITE sip:14109850123 at 64.33.1.8 SIP/2.0
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>Date: Sat, 12 Jul 2003 03:24:34 GMT
>>CSeq: 102 INVITE
>>User-Agent: CSCO/4
>>Contact: <sip:3015321510 at 128.151.224.33:5060>
>>Proxy-Authorization: Digest 
>>username="3015321510",realm="asterisk",uri="sip:64.33.1.8",response="4a9e7d0429571ec4047634179fc43f2d",nonce="2c9c06be",algorithm=md5
>>Expires: 180
>>Content-Type: application/sdp
>>Content-Length: 247
>>Remote-Party-ID: "3015321510" 
>><sip:3015321510 at 128.151.224.33>;party=calling;id-type=subscriber;privacy=off;screen=no
>>
>>v=0
>>o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33
>>s=SIP Call
>>c=IN IP4 128.151.224.33
>>t=0 0
>>m=audio 19364 RTP/AVP 0 8 18 101
>>a=rtpmap:0 PCMU/8000
>>a=rtpmap:8 PCMA/8000
>>a=rtpmap:18 G729/8000
>>a=rtpmap:101 telephone-event/8000
>>a=fmtp:101 0-15
>>
>>14 headers, 11 lines
>>Using latest request as basis request
>>Sending to 128.151.224.33 : 5060 (non-NAT)
>>Found audio format 0
>>Found audio format 8
>>Found audio format 18
>>Found audio format 101
>>Found description format PCMU
>>Found description format PCMA
>>Found description format G729
>>Found description format telephone-event
>>Capabilities: us - 2147483647, them - 268/0, combined - 268
>>Non-codec capabilities: us - 1, them - 1, combined - 1
>>Looking for 14109850123 in from-sip
>>list_route: hop: <sip:3015321510 at 128.151.224.33:5060>
>>Transmitting (no NAT):
>>SIP/2.0 100 Trying
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>;tag=as15f82296
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>CSeq: 102 INVITE
>>User-Agent: Asterisk PBX
>>Contact: <sip:14109850123 at 64.33.1.8>
>>Content-Length: 0
>>
>>
>>  to 128.151.224.33:5060
>>We're at 64.33.1.8 port 18128
>>Answering with preferred capability 2147483647
>>Answering with non-codec capability 1
>>Reliably Transmitting (no NAT):
>>SIP/2.0 200 OK
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>;tag=as15f82296
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>CSeq: 102 INVITE
>>User-Agent: Asterisk PBX
>>Contact: <sip:14109850123 at 64.33.1.8>
>>Content-Type: application/sdp
>>Content-Length: 171
>>
>>v=0
>>o=root 10711 10711 IN IP4 64.33.1.8
>>s=session
>>c=IN IP4 64.33.1.8
>>t=0 0
>>m=audio 18128 RTP/AVP 101
>>a=rtpmap:101 telephone-event/8000
>>a=fmtp:101 0-16
>>
>>  to 128.151.224.33:5060
>>     -- Playing 'demo-congrats'
>>Sip read:
>>ACK sip:14109850123 at 64.33.1.8:5060 SIP/2.0
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>;tag=as15f82296
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>Date: Sat, 12 Jul 2003 03:24:35 GMT
>>CSeq: 102 ACK
>>User-Agent: CSCO/4
>>Content-Length: 0
>>
>>
>>9 headers, 0 lines
>>Sip read:
>>BYE sip:14109850123 at 64.33.1.8:5060 SIP/2.0
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>;tag=as15f82296
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>Date: Sat, 12 Jul 2003 03:24:35 GMT
>>CSeq: 103 BYE
>>User-Agent: CSCO/4
>>Content-Length: 0
>>Proxy-Authorization: Digest 
>>username="3015321510",realm="asterisk",uri="sip:64.33.1.8",response="7cff262c42f1573c70d97968526cfdc5",nonce="2c9c06be",algorithm=md5
>>
>>
>>10 headers, 0 lines
>>Sending to 128.151.224.33 : 5060 (non-NAT)
>>Transmitting (no NAT):
>>SIP/2.0 200 OK
>>Via: SIP/2.0/UDP 128.151.224.33:5060
>>From: "3015321510" 
>><sip:3015321510 at 64.33.1.8>;tag=0002b9eb0ef4012c1a228361-11beb479
>>To: <sip:14109850123 at 64.33.1.8>;tag=as15f82296
>>Call-ID: 0002b9eb-0ef4027c-42499343-10587c26 at 128.151.224.33
>>CSeq: 103 BYE
>>User-Agent: Asterisk PBX
>>Contact: <sip:14109850123 at 64.33.1.8>
>>Content-Length: 0
>>
>>
>>  to 128.151.224.33:5060
>>
>>*CLI>
>>*CLI>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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