[asterisk-bugs] [Asterisk 0011284]: Agent transfering cal via SIP transfer gets logged out

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11284 
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Reported By:                nasirq
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11284
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.12.1  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             11-18-2007 11:07 CST
Last Modified:              11-19-2007 07:58 CST
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Summary:                    Agent transfering cal via SIP transfer gets logged
out
Description: 
Using Polycom 650 for Agent (SIP Protocol SIP/138)
Idefisk for Caller (IAX2 Protocol IAX2/nasir-lt-iax)
Cisco ATA 186 for new extension (SIP Protocol SIP/7777)

1. Agents logs in from Polycom phone using AgentLogin function
2. Caller from Idefisk gets into the Queue
3. Agents get the call
4. Agent transfers using the transfer button to extension 502 (I have
tried both attended and blond transfer)
5. Cisco ATA get the call
6. When the agent completes the transfer, the call is transfered fine,
(Idefisk and Cisco ATA can talk to each other) but the agent is logged of.

I have tried the same process by making the Idefisk the agent, and the
transfer works fine. The agent remains logged in, meaning something is
broken in SIP transfer.

Asterisk CLI output:

[Nov 18 21:58:16]     -- Executing [281002 at extensions:1]
AgentLogin("SIP/138-082340b0", "1002") in new stack
[Nov 18 21:58:16]     -- <SIP/138-082340b0> Playing 'agent-pass' (language
'en')
[Nov 18 21:58:18]     -- <SIP/138-082340b0> Playing 'agent-loginok'
(language 'en')
[Nov 18 21:58:20]     -- Started music on hold, class 'default', on
SIP/138-082340b0
[Nov 18 21:58:20]   == Agent '1002' logged in (format ulaw/ulaw)
[Nov 18 21:58:22]     -- Accepting AUTHENTICATED call from 192.168.0.199:
       > requested format = gsm,
       > requested prefs = (),
       > actual format = gsm,
       > host prefs = (g726),
       > priority = mine
[Nov 18 21:58:22]     -- Executing [2923232 at extensions:1]
Set("IAX2/nasir-lt-iax-2", "CALLERID(num)=23232") in new stack
[Nov 18 21:58:22]     -- Executing [2923232 at extensions:2]
Queue("IAX2/nasir-lt-iax-2", "support") in new stack
[Nov 18 21:58:22]     -- Started music on hold, class 'default', on
IAX2/nasir-lt-iax-2
[Nov 18 21:58:22]     -- Stopped music on hold on SIP/138-082340b0
[Nov 18 21:58:22]     -- agent_call, call to agent '1002' call on
'SIP/138-082340b0'
[Nov 18 21:58:22]     -- <SIP/138-082340b0> Playing 'beep' (language
'en')
[Nov 18 21:58:23]     -- Agent/1002 answered IAX2/nasir-lt-iax-2
[Nov 18 21:58:23]     -- <Agent/1002> Playing 'CallAnnouncement' (language
'en')
[Nov 18 21:58:26]     -- Stopped music on hold on IAX2/nasir-lt-iax-2
[Nov 18 21:58:31]     -- Started music on hold, class 'default', on
IAX2/nasir-lt-iax-2
[Nov 18 21:58:36]     -- Executing [502 at extensions:1]
Dial("SIP/138-08223910", "SIP/7777|40|whtWHT") in new stack
[Nov 18 21:58:36]     -- Called 7777
[Nov 18 21:58:36]  Extension Changed 4308 new state Ringing for Notify
User 138
[Nov 18 21:58:38]     -- SIP/7777-082497a0 is ringing
[Nov 18 21:58:39]     -- Stopped music on hold on IAX2/nasir-lt-iax-2
[Nov 18 21:58:39]     -- Started music on hold, class 'default', on
SIP/138-082340b0
[Nov 18 21:58:39]   == Spawn extension (extensions, 2923232, 2) exited
non-zero on 'SIP/138-08223910<ZOMBIE>'
[Nov 18 21:58:39]     -- Stopped music on hold on SIP/138-082340b0
[Nov 18 21:58:39]   == Agent '1002' logged out
[Nov 18 21:58:39]   == Spawn extension (extensions, 281002, 1) exited
non-zero on 'SIP/138-082340b0'
[Nov 18 21:58:43]     -- SIP/7777-082497a0 answered IAX2/nasir-lt-iax-2
[Nov 18 21:58:43]  Extension Changed 4308 new state Busy for Notify User
138
[Nov 18 21:58:47]   == Spawn extension (extensions, 502, 1) exited
non-zero on 'IAX2/nasir-lt-iax-2'
[Nov 18 21:58:47]     -- Hungup 'IAX2/nasir-lt-iax-2'


queue_log :

1195405100|1195405096.496|NONE|Agent/1002|AGENTLOGIN|SIP/138-082340b0
1195405102|1195405102.497|support|NONE|ENTERQUEUE||23232
1195405106|1195405102.497|support|Agent/1002|CONNECT|4|1195405102.498
1195405119|1195405102.497|support|Agent/1002|COMPLETECALLER|4|13|1
1195405119|1195405096.496|NONE|Agent/1002|AGENTLOGOFF|SIP/138-082340b0|19


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---------------------------------------------------------------------- 
 file - 11-19-07 07:58  
---------------------------------------------------------------------- 
oej: If they are using AgentLogin it sure can, the SIP channel itself can
queue a hangup which effectively logs off the agent. That is what is
happening here and what I (personally) expect to happen when I use both a
blind and attended transfer on a SIP phone. They both hang up once the
transfer is complete. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-19-07 07:58  file           Note Added: 0073919                          
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