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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