On 7/19/06, Dan Brummer <[EMAIL PROTECTED]> 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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