[asterisk-users] Warm transfer issues in 1.2.10

[Dan Brummer]

Setting canreinvite=no on all the sip peers and gateway made the warm
transfer work.  I'm still noticing <ZOMBIE> SIP entries, is this an
issue?

  == Spawn extension (ANC, 1691, 2) exited non-zero on
'SIP/1691-09766938<ZOMBIE>' 

Thank you,
Dan

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
Davies
Sent: Thursday, July 20, 2006 1:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Warm transfer issues in 1.2.10

On 7/19/06, Dan Brummer <dan.brummer at vegas.com> wrote:
> Hello,
> Well I was having transfer issues in 1.2.9.1 so I downgraded to
1.2.7.1.
> For testing I installed 1.2.10 on a test server and setup two Polycom 
> SIP phones.  Tried the transfer on this configuration and had the same
issues.
> Here is a log from the console:
>
> This is how the flow goes:
>
> Outside call from the PSTN to ext 1678.
> 1678 hits transfer button and dials 2175.
> 2175 answers call, speaks, then 1678 hits transfer again.
>
> After that call just goes blank and we get nothing.  Is there a fix 
> for this in the latest build?

Suggestions:

Firstly, check that there is no newer firmware for the Polycom SIP
phones - If this were a common issue in Asterisk, I expect we'd be
hearing about it a lot more...

Secondly, capture a SIP trace of what is happening - It may be related
to an attempt to negotiate codecs between the SIP phones, or numerous
other things.

Also, does it work if you set 'canreinvite=no' to keep asterisk in the
call path? Worth checking, even if it is not a setting you plan on
keeping.

Regards,
Steve
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