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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 128.151.224.33:5060
From: "3015321510" <sip:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
Date: Sat, 12 Jul 2003 03:24:34 GMT
CSeq: 101 INVITE
User-Agent: CSCO/4
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 247
Accept: application/sdp
Remote-Party-ID: "3015321510" <sip:[EMAIL
PROTECTED]>;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:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>;tag=as74174b76
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 128.151.224.33:5060
From: "3015321510" <sip:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>;tag=as74174b76
Call-ID: [EMAIL PROTECTED]
Date: Sat, 12 Jul 2003 03:24:34 GMT
CSeq: 101 ACK
Content-Length: 0
8 headers, 0 lines
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 128.151.224.33:5060
From: "3015321510" <sip:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
Date: Sat, 12 Jul 2003 03:24:34 GMT
CSeq: 102 INVITE
User-Agent: CSCO/4
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL
PROTECTED]>;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:[EMAIL PROTECTED]:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 128.151.224.33:5060
From: "3015321510" <sip:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>;tag=as15f82296
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>;tag=as15f82296
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 128.151.224.33:5060
From: "3015321510" <sip:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>;tag=as15f82296
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 128.151.224.33:5060
From: "3015321510" <sip:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>;tag=as15f82296
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=0002b9eb0ef4012c1a228361-11beb479
To: <sip:[EMAIL PROTECTED]>;tag=as15f82296
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 128.151.224.33:5060
*CLI>
*CLI>
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