[asterisk-bugs] [Asterisk 0011284]: Agent transfering cal via SIP transfer gets logged out
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Dec 18 11:49:12 CST 2007
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: 12-18-2007 11:49 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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nasirq - 12-18-07 11:49
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I tried to do an attended transfer, but I do not know how to do it.
The output for a blind transfer is:
[Dec 18 22:38:46] == Spawn extension (extensions, 302, 31) exited
non-zero on 'IAX2/nasir-lt-iax-17'
[Dec 18 22:38:46] -- Hungup 'IAX2/nasir-lt-iax-17'
[Dec 18 22:38:46] -- Executing [408 at extensions:1]
Dial("SIP/138-08222430", "SIP/7777|20|whtWHTdj") in new stack
[Dec 18 22:38:46] -- Called 7777
[Dec 18 22:38:48] -- SIP/7777-082291f8 is ringing
[Dec 18 22:38:56] -- SIP/7777-082291f8 answered SIP/138-08222430
However I have started experiencing another problem. The transfers work
fine when transferring normal calls, but when I use the IAX phone to login
as an agent, and then transfer the call, my system goes down real fast,
with load average increasing to more than 100. Killing asterisk restores
everything.
[Dec 18 22:47:58] -- Executing [296656 at extensions:7]
Queue("SIP/138-08238a90", "tollfree") in new stack
[Dec 18 22:47:58] WARNING[3767]: translate.c:163 framein: no samples for
ulawtolin
[Dec 18 22:47:58] -- Started music on hold, class 'tollfree', on
SIP/138-08238a90
[Dec 18 22:47:58] -- Stopped music on hold on IAX2/nasir-lt-iax-7
[Dec 18 22:47:58] -- agent_call, call to agent '1001' call on
'IAX2/nasir-lt-iax-7'
[Dec 18 22:47:58] -- <IAX2/nasir-lt-iax-7> Playing 'beep' (language
'en')
[Dec 18 22:47:59] -- Agent/1001 answered SIP/138-08238a90
[Dec 18 22:47:59] -- Stopped music on hold on SIP/138-08238a90
mit2*CLI>
mit2*CLI>
mit2*CLI>
mit2*CLI>
mit2*CLI>
mit2*CLI>
mit2*CLI>
[Dec 18 22:48:25] -- Started music on hold, class 'agent', on
IAX2/nasir-lt-iax-7
[Dec 18 22:48:25] == Spawn extension (extensions, 408, 0) exited
non-zero on 'SIP/138-08238a90'
[Dec 18 22:48:25] -- Executing [408 at extensions:1]
Dial("SIP/138-08238a90", "SIP/7777|20|whtWHTdj") in new stack
[Dec 18 22:48:25] -- Called 7777
[Dec 18 22:48:25] Extension Changed 4308 new state Ringing for Notify
User 138
[Dec 18 22:48:25] WARNING[3761]: interface.c:215 decodeMP3: Junk at the
beginning of frame 49443303
[Dec 18 22:48:26] -- SIP/7777-0823dcf0 is ringing
[Dec 18 22:48:37] Extension Changed 4308 new state Busy for Notify User
138
[Dec 18 22:48:37] -- SIP/7777-0823dcf0 answered SIP/138-08238a90
Sometimes the transfer goes ok, and when that happens, the agent on the
IAX phone remains logged in, ready to take another call.
Issue History
Date Modified Username Field Change
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12-18-07 11:49 nasirq Note Added: 0075656
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