[asterisk-bugs] [JIRA] (ASTERISK-26414) app_externalivr: ExternalIVR attended transfer - no audio

[balamurugan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26414?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234857#comment-234857 ] 

balamurugan commented on ASTERISK-26414:
----------------------------------------

FYI , 

 From the asterisk log you provided 

I do see before Tranfer initiated there is a call from below anyidea who is this and why a call from 192.168.7.31

<--- SIP read from UDP:192.168.7.31:15060 --->
INVITE sip:125 at 172.16.16.91 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.31:15060;rport;branch=z9hG4bKlrbmvtet
Max-Forwards: 70
To: <sip:125 at 172.16.16.91>
From: "202" <sip:202 at 172.16.16.91>;tag=kawpp
Call-ID: alspjympmhyetvt at debian-one
CSeq: 746 INVITE
Contact: <sip:202 at 192.168.7.31:15060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Subject: Call transfer - 201 <sip:201 at 172.16.16.91>
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.4.2
Content-Length: 307

Followed by  there is Transfer to Replace with above call . Need to understand what exactly the Called user is doing 
with REFER from Called user .

Also you shared log from asterisk 13.11.2 
Can you share the working log ??

thanks,
bala


> app_externalivr: ExternalIVR attended transfer - no audio
> ---------------------------------------------------------
>
>                 Key: ASTERISK-26414
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26414
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_externalivr
>    Affects Versions: 11.23.1, 13.11.2
>         Environment: Debian 7 64bit
>            Reporter: Chris Maciejewski
>            Assignee: Unassigned
>         Attachments: asterisk.log, call.log, call.pcap
>
>
> Hi,
> When attended transfer to ExternalIVR is completed *after* ExternalIVR app starts no audio is sent to SIP client.
> To reproduce the bug:
> 1. Download example config which can be found here:
> https://github.com/level7systems/asterisk-cfg/tree/ExternalIVR-xfer
> 2. Register SIP client for ext 201 and ext 202.
> 3. Perform call scenario as below:
> a) 201 calls 202
> b) 202 performs attended transfer to ext 125 (ExternalIVR). 202 waits 10 seconds before completing the transfer
> c) when transfer is completed there is no audio sent to 201
> Note: if the same scenario is performed, however in step b) 202 waits only 3 seconds before completing the transfer audio is sent to 201 correctly. So it seems the issue doesn't occur if attended transfer is completed *before* ExternalIVR app starts and happens only if transfer completes when ExternalIVR app already started.
> Note 2: the same problem affects ver. 11.22.0 as well.
> Regards,
> Chris



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