[Asterisk-Users] SIP immediate hangups with latest CVS

Mon Jul 14 12:18:23 MST 2003

The resolution to this problem was provided by Martin @Digium, who 
said that there must be at least one "or" in the codec permissions 
line.  In other words, I have "allow=all" at the top of the sip.conf 
file, but I should have something like:

disallow=all
allow=alaw

Asterisk expects an ordering of some type to be defined with "or" 
statements, which is a bit confusing, but I suppose it makes logical 
sense.  An easier way would have been to leave out the "allow=" line 
entirely.

JT

>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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