Open source replacement for AudioCodes nCite 1000 SBC

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List users, I have an AudioCodes nCite 1000 SBC that is end-of-life and I'm
looking to replace it with open source software. I believe one of the
SIP proxy projects will fit my needs, but I'm a bit overwhelmed by
the number of choices and I'd like the advice of experienced users
before I venture too far down any one path. The projects that came to mind first were Kamailio, OpenSIPS, SER, and
SIP-Router, but I'm aware that there are others and I'm open to
suggestions. Please keep in mind that I'm looking for…

Asterisk Users 3.2 years ago 0 Answer

Asterisk - Nortel transfer problem

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Hi Carlos It could help if you can get a trace of the call transfer from Nortel to SIP extension on the Asterisk (1303), if no way to get from Nortel get from Asterisk.
I guest operator try to make a bind call transfer, without wait complete DR2 signalling exchange at least minimal time exchange DR2 signalling between Nortel and Asterisk is about 5 sec.
Best regards Mc GRATH Ricardo
E-Mail mcgrathr@mail2web.com

Asterisk Users 3.2 years ago 0 Answer

CONNECTEDLINE() updated during SIP events?

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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

Asterisk Users 3.2 years ago 3 Answer