Hi,
Is it possible yet to restart a single Dahdi span in any version of
Asterisk? (instead of all of them)

Thanks.

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 something
reasonably easy to setup and administer. I’m not looking to have it
setup tomorrow, but it must be something that a single skilled
Linux/Asterisk administrator could take on in addition to other daily
tasks.

The functionality that I’m currently using on the nCite 1000 is:

SIP Proxy/B2BUA and RTP Proxy
* Internal call routing (private IP-to-private IP)
* External call routing (external IP-to-private IP and vice versa)
with topology hiding
* SIP header modification
* Digit manipulation (delete digits/add prefixes based on matching
criteria)

Connectivity
* NAT traversal
* External registrations (registration bindings are maintained and
ports on the far end firewall are kept open)

Authentication
* By source IP address or range
* By destination SIP proxy

Session Targets and Session Target Sets
* Individual SIP entities (e.g. Asterisk servers, SIP trunks) are
defined as session targets
* Session targets are grouped into sets with call distribution based
on priorities/weights

Call Routing
* Static Binding: All calls to an inbound SIP proxy are routed to
the same session target set via the same outbound SIP proxy
* Dial Pattern: All calls to an inbound SIP proxy are routed to
different session target sets via different outbound SIP proxies
based on dial patterns

Future Considerations
* TLS/SRTP support

Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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

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