[Asterisk-Users] SIP immediate hangups with latest CVS
John Todd
jtodd at loligo.com
Fri Jul 11 20:42:14 MST 2003
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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