[asterisk-users] Warm transfer issues in 1.2.10

[Steve Davies]

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