[Asterisk-Users] SIP immediate hangups with latest CVS

Fri Jul 11 20:57:07 MST 2003

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>
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