Hi,

I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:

- Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a SIP divert is in place? If so, how?

- Is it possible to have the COLP/COLR information updated when a SIP
attended transfer is completed? If so how?

In both of the above cases, there is no obvious dialplan execution
when the calls are redirected, diverted or masqueraded, so we cannot
update the CONNECTEDLINE() information trivially. Or am I missing an
obvious trick?

Thanks,
Steve

I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?

Thanks

Bryant

Hello,

is there a way to put a certain SIP peer on state “busy” ?

I know you can do this by pressing “DND” on your IP-phone, but can this
state also be set in the dialplan ?

Thanks.

Jonas.

Hi, I am encountering problem making concurrent calls using A sangoma card, It seems that the 2nd call get a congested or buzy,I connect via sip–>asterisk–>dahdi attached is the PRI debug messages

Can i don’t sent into the SIP invite the “Session Timer” ? on asterisk 1.6

Best regards
Olivier