[asterisk-users] Call transfer issues

[Kevin Smith]

My guess is I stumped everyone ;)

Anyway, I rolled back asterisk to 1.2.9.1 (same for libpri and zaptel 
back one release) and transfers were working again. Now I'm still quite 
new to asterisks, I know enough to hold my own, but not enough to know 
the full inter workings of it. But here is my thought:

Caller A calls in and talks to Employee B. B wants to transfer to C. 
Asterisk sets up the bridge between B and C. B completes the transfer. 
Now A and C are connected but there is no audio stream. If C or A puts 
the other on hold, and then resumes the call, audio is restored.

By that I would say placing them on hold clears a flag or updates one to 
connect the audio stream? Or am I way off on this assumption? Also if 
this sounds like a possible bug, what information do I need to include, 
or is good to include, when submitting bugs?

Thanks,
Kevin

Kevin Smith wrote:
> Hey everyone,
>
> Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk 
> 1.2.10. It has been reported to me when doing an attended transfer the 
> audio drops out. I ran a few different tests and here is what I noticed.
>
> 1. Blind transfers work with no problem.
> 2. Attended transfers were you transfer the call before the person 
> picks up works.
> 3. If the person the call is being transferred to answers and then the 
> transfer completes, the audio drops.
>
> I noticed in the CLI the following (I replaced the number with XXXXXXX's)
>
> -- Attempting native bridge of SIP/989XXXXXXX-b76167c8 and 
> SIP/989XXXXXXX-08f956b8
>  == Parsing '/etc/asterisk/manager.conf': Found
>    -- Stopped music on hold on Zap/2-1
>  == Spawn extension (Mercury-Directory-Dialer, 989XXXXXXX, 8) exited 
> non-zero on 'SIP/989XXXXXXX-b76167c8<ZOMBIE>'
>    -- Executing Hangup("SIP/989XXXXXXX-b76167c8<ZOMBIE>", "") in new 
> stack
>  == Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero 
> on 'SIP/989XXXXXXX-b76167c8<ZOMBIE>'
>    -- Incoming call: Got SIP response 500 "Internal Server Error" back 
> from 64.7.177.103
>
> Now what I noticed is that once the transfer is done, I'm still 
> connected the the person that called me to do an attended transfer. 
> However, if I hang up the phone, the call drops. If I place the call 
> on hold and take them off hold, audio is resumed and everything works 
> normally.
>
> Here is the conf information
>
> exten => s,1,SetCallerID(${ARG1})
> exten => s,n,Set(DST_EXT_NUM=${ARG2})
> exten => s,n,gotoif,$[${ARG2}=989XXXXXX]?TIME:GOON     ;Add test if 
> hours is the basis for voice mail
>
> exten => s,n(GOON),AGI(VoiceMail.php)   ;Test for phone status
> exten => s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
> exten => s,n,Dial(SIP/${ARG2},25)
>
> ...VoiceMail choice....
>
> exten => h,1,HangUp()
>
> Where I have VoiceMail choice it takes the variables from the AGI 
> script and decides which voice message to play. But the problem is 
> happening before that occurs so I don't think it has anything to do 
> with the problem.
>
> Any ideas to what could be the cause or how to correct it? SIP version 
> or does the new asterisk build have any new features enabled by 
> default that the older build would not? Any suggestions or thoughts 
> would be greatly helpful.
>
> Kevin
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