[Asterisk-Users] No audio in either direction, sip channels hanging, asterisk will
not shut down.
Rhys Hopkins
rhysh at aerotech-eu.net
Thu Aug 21 03:56:53 MST 2003
Hi all,
I have been asked to look into using asterisk as part of our setup.
The eventual goal is to replace as many parts of the existing setup as
possible, but in the interim, I just have to make it bolt on and work
with all existing parts.
My current setup is as follows:
Cisco 7940
(ext 2000)
|
v
Asterisk -> Snom SIP proxy(v2.22) -> Vega100 PSTN gw -> Index PBX
| | |
v v v
Cisco 7940 Cisco 7940 Phone
(ext 2001) (ext 7038) (ext. 3046)
All the sip connections are working fine, i.e. I can call between
extensions 2000, 2001, 7038 on the above diagram with no problems.
Also the connection from the Snom Proxy to the PBX works fine.
(7038->3046) and vice versa.
The problem I have is that I can ring from the phones on asterisk to the
phones on the PBX, but I have no audio in either direction.
After attempting this the call seems to not terminate properly, the sip
phone involved refuses to cooperate, and asterisk refuses to shut down.
If I issue "stop now", the CLI just hangs indefinitely. "sip show
channels" reports 2 active channels ( between the phones involved ).
Firstly, is there any way I can rescue asterisk without doing a "killall
-9 asterisk", which is how I am currently dealing with this. ( Also, how
does one determine the <channel> parameter to be used with "soft hangup" ?)
Secondly, I have attached the trace from the CLI with "sip debug" turned
on. Does this shed any light on what is causing the problem ?
I appreciate this is an involved query, but I have been poring over the
sip debug trace for a whole day now, and it still makes no sense to me.
Any help would be greatly appreciated.
Regards,
Rhys.
-------------- next part --------------
-- Executing Dial("SIP/2001-9180", "SIP/3046 at sip.culver-tec.com|20") in new stack
We're at 62.254.245.18 port 13270
Answering with preferred capability 4
Answering with preferred capability 8
Answering with capability 2
Answering with non-codec capability 1
11 headers, 11 lines
Reliably Transmitting:
INVITE sip:3046 at sip.culver-tec.com SIP/2.0
Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848
From: "2001" <sip:2001 at 62.254.245.18>;tag=as6992a8b0
To: <sip:3046 at sip.culver-tec.com>
Contact: <sip:2001 at 62.254.245.18>
Call-ID: 171697661abf036a2b3fa6f8741d4749 at 62.254.245.18
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 19457 19457 IN IP4 62.254.245.18
s=session
c=IN IP4 62.254.245.18
t=0 0
m=audio 13270 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 62.254.245.14:5060
-- Called 3046 at sip.culver-tec.com
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848
From: "2001" <sip:2001 at 62.254.245.18>;tag=as6992a8b0
To: <sip:3046 at sip.culver-tec.com>
Call-ID: 171697661abf036a2b3fa6f8741d4749 at 62.254.245.18
CSeq: 102 INVITE
Content-Length: 0
7 headers, 0 lines
We're at 62.254.245.18 port 17826
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
Transmitting (no NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.101.186:5060
From: "2001" <sip:2001 at asterisk.culver-tec.com>;tag=000ab71451fd000d6dda63bb-533718c4
To: <sip:3046 at asterisk.culver-tec.com>;tag=as5fa023e1
Call-ID: 000ab714-51fd0010-6f1d44d7-6ca818fe at 192.168.101.186
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3046 at 62.254.245.18>
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 19457 19457 IN IP4 62.254.245.18
s=session
c=IN IP4 62.254.245.18
t=0 0
m=audio 17826 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 192.168.101.186:5060
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848
Record-route: <sip:62.254.245.14:5060;lr=1>
From: "2001" <sip:2001 at 62.254.245.18>;tag=as6992a8b0
To: <sip:3046 at sip.culver-tec.com>
Call-ID: 171697661abf036a2b3fa6f8741d4749 at 62.254.245.18
CSeq: 102 INVITE
Contact: <sip:3046 at vega.culver-tec.com:5060;maddr=62.254.245.12>
User-Agent: Vega100-T1E1/08.02.05.1xT017
Content-Type: application/sdp
Content-Length: 190
v=0
o=Vega50 11 1 IN IP4 62.254.245.12
s=Sip Call
t=0 0
m=audio 10014 RTP/AVP 0 101
c=IN IP4 62.254.245.12
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
11 headers, 9 lines
-- SIP/sip.culver-tec.com-aa40 is ringing
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848
Record-route: <sip:62.254.245.14:5060;lr=1>
From: "2001" <sip:2001 at 62.254.245.18>;tag=as6992a8b0
To: <sip:3046 at sip.culver-tec.com>
Call-ID: 171697661abf036a2b3fa6f8741d4749 at 62.254.245.18
CSeq: 102 INVITE
Contact: <sip:3046 at vega.culver-tec.com:5060;maddr=62.254.245.12>
User-Agent: Vega100-T1E1/08.02.05.1xT017
Content-Type: application/sdp
Content-Length: 190
v=0
o=Vega50 11 1 IN IP4 62.254.245.12
s=Sip Call
t=0 0
m=audio 10014 RTP/AVP 0 101
c=IN IP4 62.254.245.12
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
11 headers, 9 lines
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848
Record-route: <sip:62.254.245.14:5060;lr=1>
From: "2001" <sip:2001 at 62.254.245.18>;tag=as6992a8b0
To: <sip:3046 at sip.culver-tec.com>
Call-ID: 171697661abf036a2b3fa6f8741d4749 at 62.254.245.18
CSeq: 102 INVITE
Contact: <sip:3046 at vega.culver-tec.com:5060;maddr=62.254.245.12>
User-Agent: Vega100-T1E1/08.02.05.1xT017
Content-Type: application/sdp
Content-Length: 190
v=0
o=Vega50 11 1 IN IP4 62.254.245.12
s=Sip Call
t=0 0
m=audio 10014 RTP/AVP 0 101
c=IN IP4 62.254.245.12
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
11 headers, 9 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848
Record-route: <sip:62.254.245.14:5060;lr=1>
From: "2001" <sip:2001 at 62.254.245.18>;tag=as6992a8b0
To: <sip:3046 at sip.culver-tec.com>;tag=0002-000C-4120B080
Call-ID: 171697661abf036a2b3fa6f8741d4749 at 62.254.245.18
CSeq: 102 INVITE
Contact: <sip:3046 at vega.culver-tec.com:5060;maddr=62.254.245.12>
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER
Accept: application/sdp, application/none
Accept-Language: en
User-Agent: Vega100-T1E1/08.02.05.1xT017
Content-Type: application/sdp
Content-Length: 190
v=0
o=Vega50 12 1 IN IP4 62.254.245.12
s=Sip Call
t=0 0
m=audio 10014 RTP/AVP 0 101
c=IN IP4 62.254.245.12
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
14 headers, 9 lines
Found audio format 0
Found audio format 101
Found description format PCMU
Found description format telephone-event
Capabilities: us - 524302, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: <sip:62.254.245.14:5060;lr=1>
list_route: hop: <sip:3046 at vega.culver-tec.com:5060;maddr=62.254.245.12>
set_destination: Parsing <sip:62.254.245.14:5060;lr=1> for address/port to send to
set_destination: set destination to 62.254.245.14, port 5060
Transmitting:
ACK sip:3046 at sip.culver-tec.com SIP/2.0
Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848
Route: <sip:3046 at vega.culver-tec.com:5060;maddr=62.254.245.12>
From: "2001" <sip:2001 at 62.254.245.18>;tag=as6992a8b0
To: <sip:3046 at sip.culver-tec.com>;tag=0002-000C-4120B080
Contact: <sip:2001 at 62.254.245.18>
Call-ID: 171697661abf036a2b3fa6f8741d4749 at 62.254.245.18
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 62.254.245.14:5060
Sip read:
CANCEL sip:3046 at asterisk.culver-tec.com SIP/2.0
Via: SIP/2.0/UDP 192.168.101.186:5060
From: "2001" <sip:2001 at asterisk.culver-tec.com>;tag=000ab71451fd000d6dda63bb-533718c4
To: <sip:3046 at asterisk.culver-tec.com>
Call-ID: 000ab714-51fd0010-6f1d44d7-6ca818fe at 192.168.101.186
Date: Thu, 21 Aug 2003 09:57:41 GMT
CSeq: 102 CANCEL
User-Agent: CSCO/4
Content-Length: 0
Proxy-Authorization: Digest username="2001",realm="asterisk",uri="sip:62.254.245.18",response="466777cb9c995d5c862fc5310d33477d",nonce="09f50874",algorithm=md5
10 headers, 0 lines
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format
62.254.245.14 3046 171697661ab 00102/00000 00000ms 0000ms 4
192.168.101.186 2001 000ab714-51 00101/00103 00000ms 0000ms 4
2 active SIP channel(s)
*CLI>
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