[asterisk-bugs] [JIRA] (ASTERISK-21172) One way audio when external Call forwarded to queue member
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Mon Jul 15 16:10:03 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21172?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=207984#comment-207984 ]
Matt Jordan commented on ASTERISK-21172:
----------------------------------------
Unfortunately, I'm not able to reproduce this problem.
I did the following to reproduce this issue:
* Three phones - {{digium01}}, {{digium02}}, and {{phoneb}} - were used. {{phoneb}} is the caller; {{digium01}} is the SIP phone that forwards the call; {{digium02}} is the queue member
* {{phoneb}} dials extension 1000, which dials {{digium01}}
* {{digium01}} forwards the request to extension 5000
* Extension 5000 puts {{phoneb}} into the {{sales}} queue
* A Local channel member dials {{digium02}}
* {{digium02}} Answers
* Both {{phoneb}} and {{digium02}} can converse. There were no one-way audio issues.
Sample configurations:
{noformat:title=extensions.conf}
[default]
exten => 1000,1,NoOp()
same => n,Verbose(1, Dialing digium01)
same => n,GoSub(default,set_caller_id,1)
same => n,Dial(SIP/digium01,15,hHkKtT)
same => n,Hangup()
exten => 5000,1,NoOp()
same => n,Queue(sales)
[agents]
exten => digium02,hint,SIP/digium02
exten => digium02,1,NoOp()
same => n,Dial(SIP/digium02,15,hHkKtT)
same => n,Hangup()
{noformat}
{noformat:title=sip.conf}
[peer-template](!)
directmedia = yes
disallow = all
allow = g722
allow = gsm
allow = ulaw
allow = alaw
[phone_b](peer-template)
type = peer
secret =
directmedia = no
callerid = "Phone B" <B>
host = dynamic
context = default
sendrpid=pai
mailbox=3000
[digium02](peer-template)
type = peer
secret = digium02
directmedia = no
callerid = "Digium 02" <102>
host = dynamic
context = default
sendrpid=pai
setvar=AGENTNAME=digium02
mailbox=2000
[digium01](peer-template)
type = peer
secret = digium01
directmedia = no
callerid = "Digium 01" <101>
host = dynamic
context = default
sendrpid=pai
mailbox=1000
setvar=AGENTNAME=digium01
{noformat}
{noformat:title=queues.conf}
[sales]
strategy=ringall
announce=sales
musicclass = default
monitor-type=MixMonitor
monitor-format=wav
member => Local/digium02 at agents
{noformat}
Below is a dump of the CLI during execution:
{noformat}
-- Executing [1000 at default:1] NoOp("SIP/phone_b-00000003", "") in new stack
-- Executing [1000 at default:2] Verbose("SIP/phone_b-00000003", "1, Dialing digium01") in new stack
Dialing digium01
-- Executing [1000 at default:3] Gosub("SIP/phone_b-00000003", "default,set_caller_id,1") in new stack
-- Executing [set_caller_id at default:1] NoOp("SIP/phone_b-00000003", "") in new stack
-- Executing [set_caller_id at default:2] Set("SIP/phone_b-00000003", "CALLERID(name)=Asterisk") in new stack
-- Executing [set_caller_id at default:3] Set("SIP/phone_b-00000003", "CALLERID(num)=555-5555") in new stack
-- Executing [set_caller_id at default:4] Return("SIP/phone_b-00000003", "") in new stack
-- Executing [1000 at default:4] Dial("SIP/phone_b-00000003", "SIP/digium01,15,hHkKtT") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/digium01
-- Got SIP response 302 "Moved Temporarily" back from 10.24.19.80:5060
-- Now forwarding SIP/phone_b-00000003 to 'Local/5000 at default' (thanks to SIP/digium01-00000004)
[Jul 15 15:51:15] NOTICE[17699]: app_dial.c:901 do_forward: Not accepting call completion offers from call-forward recipient Local/5000 at default-00000003;1
-- Executing [5000 at default:1] NoOp("Local/5000 at default-00000003;2", "") in new stack
-- Executing [5000 at default:2] Queue("Local/5000 at default-00000003;2", "sales") in new stack
-- Started music on hold, class 'default', on Local/5000 at default-00000003;2
-- Executing [digium02 at agents:1] NoOp("Local/digium02 at agents-00000004;2", "") in new stack
-- Executing [digium02 at agents:2] Dial("Local/digium02 at agents-00000004;2", "SIP/digium02,15,hHkKtT") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/digium02
-- Local/digium02 at agents-00000004;1 connected line has changed. Saving it until answer for Local/5000 at default-00000003;2
-- Local/digium02 at agents-00000004;1 connected line has changed. Saving it until answer for Local/5000 at default-00000003;2
-- SIP/digium02-00000005 is ringing
-- Local/digium02 at agents-00000004;1 is ringing
-- Local/digium02 at agents-00000004;1 connected line has changed. Saving it until answer for Local/5000 at default-00000003;2
-- SIP/digium02-00000005 answered Local/digium02 at agents-00000004;2
-- Local/digium02 at agents-00000004;1 answered Local/5000 at default-00000003;2
-- Stopped music on hold on Local/5000 at default-00000003;2
== Begin MixMonitor Recording Local/5000 at default-00000003;2
-- Local/5000 at default-00000003;1 answered SIP/phone_b-00000003
-- Executing [h at default:1] NoOp("Local/5000 at default-00000003;2", "") in new stack
-- Executing [h at default:2] Wait("Local/5000 at default-00000003;2", "10") in new stack
== Spawn extension (default, h, 2) exited non-zero on 'Local/5000 at default-00000003;2'
== Spawn extension (default, 5000, 2) exited non-zero on 'Local/5000 at default-00000003;2'
== Spawn extension (agents, digium02, 2) exited non-zero on 'Local/digium02 at agents-00000004;2'
-- Executing [h at default:1] NoOp("SIP/phone_b-00000003", "") in new stack
-- Executing [h at default:2] Wait("SIP/phone_b-00000003", "10") in new stack
== Spawn extension (default, h, 2) exited non-zero on 'SIP/phone_b-00000003'
== MixMonitor close filestream
== End MixMonitor Recording Local/5000 at default-00000003;2
== Spawn extension (default, 1000, 4) exited non-zero on 'SIP/phone_b-00000003'
{noformat}
I'm not sure what else we can do on this issue. We cannot reproduce it, nor can we determine from your pcaps what the issue is. As Rusty mentioned, in the {{no-audio.pcap}} RTP appears to be flowing at all times between all three participants. From Asterisk's perspective, it appears as if RTP is going back and forth between the phones.
Since we can't reproduce this issue, I'm going to go ahead and close it out as "Can't Reproduce". If you can determine what in the configuration is different between my attempt at reproducing the problem and your lab set up and can suggest how we can reproduce the problem, we can reopen this issue.
> One way audio when external Call forwarded to queue member
> ----------------------------------------------------------
>
> Key: ASTERISK-21172
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-21172
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Applications/app_queue, Channels/chan_local, Channels/chan_sip/General
> Affects Versions: SVN, 1.8.20.1, 1.8.21.0
> Environment: Ubuntu 10.04LTS, I686pae
> Reporter: Peter Katzmann
> Assignee: Matt Jordan
> Attachments: digium-info_20130226.tar.gz, flow-queue, flow-user, from-asterisk-to-queue1817.cap, from-asterisk-to-queuemember-1817, from-asterisk-to-queue-member.pcap.bz2, from-ext-to-asterisk.pcap.bz2, from-patton-to-asterisk-1817, from-patton-to-asterisk-1817-2.pcap, logs-1921-v2-patch.tar.bz, no-audio.log, no-audio.pcap, oneWayAudio-1.8.21, onewayaudio-1.8.21.pcap, oneWayAudio-svn, onewayaudio-svn.pcap, pcaps-1921-v2-patch.tar.bz, withandwithout.pcap
>
>
> When i call from external an colleague and he is not available the call is forwarded to the hotline queue.
> On queue operator pick up the call, ringing stops on caller side ba he can't here the agent but the agent can hear the caller.
> When i try the same scenario internal audio ist ok
> Asterisk 1.8.17 is working fine
> Core dump during one call is available so if there is some information inside give me some information how to retrieve them.
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