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Summary:ASTERISK-07997: Called side still ringing when calling side hangup
Reporter:Pavel Yarmolchuk (xpasha)Labels:
Date Opened:2006-10-24 16:07:28Date Closed:2011-06-07 14:03:01
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Called side still ringing when calling side hangup.
This example:

[default]
exten => 997960,1,Dial(SIP/084322997828@217.x.x.1|10)

When caller hangs up not waiting answer, phone 997828 still ringing. Both sides are Cisco AS5350.

****** ADDITIONAL INFORMATION ******

*CLI> sip debug
SIP Debugging re-enabled
*CLI>
<--- SIP read from 217.x.x.1:53008 --->
INVITE sip:997960@217.x.x.9:5060 SIP/2.0
Via: SIP/2.0/UDP  217.x.x.1:5060
From: <sip:9047636845@217.x.x.1>;tag=1E443FDC-100E
To: <sip:997960@217.x.x.9>
Date: Tue, 24 Oct 2006 21:04:23 GMT
Call-ID: EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 250743114-1658458587-2168581809-349256015
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:9047636845@217.x.x.1>;party=calling;screen=yes;privacy=off
Timestamp: 1161723863
Contact: <sip:9047636845@217.x.x.1:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 6953 6845 IN IP4 217.x.x.1
s=SIP Call
c=IN IP4 217.x.x.1
t=0 0
m=audio 17696 RTP/AVP 8 101
c=IN IP4 217.x.x.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (20 headers 11 lines) ---
Sending to 217.x.x.1 : 5060 (no NAT)
Using INVITE request as basis request - EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
Found no matching peer or user for '217.x.x.1:53008'
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 217.x.x.1:17696
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 217.x.x.1:17696
Looking for 997960 in default (domain 217.x.x.9)
list_route: hop: <sip:9047636845@217.x.x.1:5060>

<--- Transmitting (no NAT) to 217.x.x.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  217.x.x.1:5060;received=217.x.x.1
From: <sip:9047636845@217.x.x.1>;tag=1E443FDC-100E
To: <sip:997960@217.x.x.9>
Call-ID: EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:997960@217.x.x.9>
Content-Length: 0


<------------>
   -- Executing [997960@default:1] Dial("SIP/217.x.x.1-081e9178", "SIP/084322997828@217.x.x.2|10") in new stack
Audio is at 217.x.x.9 port 11830
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.x.x.2:5060:
INVITE sip:084322997828@217.x.x.2 SIP/2.0
Via: SIP/2.0/UDP 217.x.x.9:5060;branch=z9hG4bK02e6391e;rport
From: "9047636845" <sip:9047636845@217.x.x.9>;tag=as2dcfd960
To: <sip:084322997828@217.x.x.2>
Contact: <sip:9047636845@217.x.x.9>
Call-ID: 0d513f893149a3425c2adf7a3f4266b2@217.x.x.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Oct 2006 21:04:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 31470 31470 IN IP4 217.x.x.1
s=session
c=IN IP4 217.x.x.1
t=0 0
m=audio 17696 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
   -- Called 084322997828@217.x.x.2

<--- SIP read from 217.x.x.2:54129 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.x.x.9:5060;branch=z9hG4bK02e6391e;rport
From: "9047636845" <sip:9047636845@217.x.x.9>;tag=as2dcfd960
To: <sip:084322997828@217.x.x.2>;tag=DCCE42D0-1E
Date: Sun, 02 Apr 2000 14:04:27 GMT
Call-ID: 0d513f893149a3425c2adf7a3f4266b2@217.x.x.9
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from 217.x.x.2:54129 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 217.x.x.9:5060;branch=z9hG4bK02e6391e;rport
From: "9047636845" <sip:9047636845@217.x.x.9>;tag=as2dcfd960
To: <sip:084322997828@217.x.x.2>;tag=DCCE42D0-1E
Date: Sun, 02 Apr 2000 14:04:27 GMT
Call-ID: 0d513f893149a3425c2adf7a3f4266b2@217.x.x.9
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:084322997828@217.x.x.2:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 238

v=0
o=CiscoSystemsSIP-GW-UserAgent 1062 5247 IN IP4 217.x.x.2
s=SIP Call
c=IN IP4 217.x.x.2
t=0 0
m=audio 17228 RTP/AVP 8 101
c=IN IP4 217.x.x.2
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (14 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 217.x.x.2:17228
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 217.x.x.2:17228
   -- SIP/217.x.x.2-081ee028 is making progress passing it to SIP/217.x.x.1-081e9178
Audio is at 217.x.x.9 port 10368
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 217.x.x.1:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP  217.x.x.1:5060;received=217.x.x.1
From: <sip:9047636845@217.x.x.1>;tag=1E443FDC-100E
To: <sip:997960@217.x.x.9>;tag=as304cef66
Call-ID: EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:997960@217.x.x.9>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 31470 31470 IN IP4 217.x.x.2
s=session
c=IN IP4 217.x.x.2
t=0 0
m=audio 17228 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

<------------>

<--- SIP read from 217.x.x.2:54129 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 217.x.x.9:5060;branch=z9hG4bK02e6391e;rport
From: "9047636845" <sip:9047636845@217.x.x.9>;tag=as2dcfd960
To: <sip:084322997828@217.x.x.2>;tag=DCCE42D0-1E
Date: Sun, 02 Apr 2000 14:04:27 GMT
Call-ID: 0d513f893149a3425c2adf7a3f4266b2@217.x.x.9
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:084322997828@217.x.x.2:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 238

v=0
o=CiscoSystemsSIP-GW-UserAgent 1062 5247 IN IP4 217.x.x.2
s=SIP Call
c=IN IP4 217.x.x.2
t=0 0
m=audio 17228 RTP/AVP 8 101
c=IN IP4 217.x.x.2
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (14 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 217.x.x.2:17228
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 217.x.x.2:17228
   -- SIP/217.x.x.2-081ee028 is making progress passing it to SIP/217.x.x.1-081e9178

<--- SIP read from 217.x.x.1:53008 --->
CANCEL sip:997960@217.x.x.9:5060 SIP/2.0
Via: SIP/2.0/UDP  217.x.x.1:5060
From: <sip:9047636845@217.x.x.1>;tag=1E443FDC-100E
To: <sip:997960@217.x.x.9>
Date: Tue, 24 Oct 2006 21:04:23 GMT
Call-ID: EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
CSeq: 101 CANCEL
Max-Forwards: 6
Timestamp: 1161723872
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 217.x.x.1 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 217.x.x.1:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP  217.x.x.1:5060;received=217.x.x.1
From: <sip:9047636845@217.x.x.1>;tag=1E443FDC-100E
To: <sip:997960@217.x.x.9>;tag=as304cef66
Call-ID: EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 217.x.x.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP  217.x.x.1:5060;received=217.x.x.1
From: <sip:9047636845@217.x.x.1>;tag=1E443FDC-100E
To: <sip:997960@217.x.x.9>;tag=as304cef66
Call-ID: EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:997960@217.x.x.9>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0d513f893149a3425c2adf7a3f4266b2@217.x.x.9' in 32000 ms (Method: INVITE)
 == Spawn extension (default, 997960, 1) exited non-zero on 'SIP/217.x.x.1-081e9178'

<--- SIP read from 217.x.x.1:53008 --->
ACK sip:997960@217.x.x.9:5060 SIP/2.0
Via: SIP/2.0/UDP  217.x.x.1:5060
From: <sip:9047636845@217.x.x.1>;tag=1E443FDC-100E
To: <sip:997960@217.x.x.9>;tag=as304cef66
Date: Tue, 24 Oct 2006 21:04:23 GMT
Call-ID: EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'EF2A60B-62DA11DB-8144EEB1-14D1394F@217.x.x.1' Method: ACK
Comments:By: Pavel Yarmolchuk (xpasha) 2006-10-24 17:44:31

obscene and inflamatory wording, as well as conclusions unsupported by factual information removed by serge-v.

Now, as to the quality of this bug report; what kind of Asterisk user:
1) files a chan_sip bug as "unsupported" channel driver?
2) posts debugging information inline, not as an attachment contrary to the bug posting guidelines?


By: Tilghman Lesher (tilghman) 2006-10-25 00:02:57

In English?
By: Anthony LaMantia (alamantia) 2006-10-25 21:28:51

Can you please provide me the IP addresses of your two AS5350 machines.
as well as the ip address of your asterisk machine,

we need this information to determine if there is a problem with asterisk and if so where it is.. and without this information this is not possible.
By: Serge Vecher (serge-v) 2006-10-26 10:42:33

As per bug guidelines, you need to attach a SIP _debug_ trace illustrating the problem. Please do the following:
1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Save complete console log to file and _attach_ said file to the bug.
 i.e. asterisk -Tvvvvvdddddgc | tee /tmp/sipdebug.txt
By: Olle Johansson (oej) 2006-11-07 11:36:14.000-0600

No answer from reporter... Will happily reopen when there's activity.