[asterisk-bugs] [Asterisk 0011284]: Agent transfering cal via SIP transfer gets logged out
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu Feb 14 14:11:35 CST 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=11284
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Reported By: nasirq
Assigned To: file
======================================================================
Project: Asterisk
Issue ID: 11284
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: major
Priority: normal
Status: resolved
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:
Resolution: not fixable
Fixed in Version:
======================================================================
Date Submitted: 11-18-2007 11:07 CST
Last Modified: 02-14-2008 14:11 CST
======================================================================
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
======================================================================
----------------------------------------------------------------------
file - 02-14-08 14:11
----------------------------------------------------------------------
After examining this further there is nothing we can do to stop this. Once
the SIP phone completes the transfer it terminates the call.
Issue History
Date Modified Username Field Change
======================================================================
02-14-08 14:11 file Status new => resolved
02-14-08 14:11 file Resolution open => not fixable
02-14-08 14:11 file Assigned To => file
02-14-08 14:11 file Note Added: 0082269
======================================================================
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