Hey everyone, the carrier I am working with for my sip trunk has some problems with Asterisk. They think since they have tried once before and failed with a Digium that they are going to have the same problems with the PIAF system that I am setting up. They have a Metaswitch at their location and they say Asterisk does not pass interop on that switch because there are so many different versions of Asterisk.
He asked me for a Approved certified Media Gateway Model he said if he had that we might be able to get it working.
Can anyone please help me counter this information or give me the proper media gateway model number for this sip provider?
Metaswitch is not a SIP provider, it’s a brand of switch.
So you want us to tell you a brand of media gateway so you can lie?
I am confused, Asterisk works fine with Metaswitch you just have to know what you are doing.
If you have to fake out the carrier you are going to have to change the SIP-UA settings to match the device you are trying to spoof.
You better also must be very fluent in SIP terminology and be able to adjust Asterisk settings to match what the provider asks you to do.
In my carrier I have never failed to get Asterisk to work with another SIP UA. I have failed on a few H.323 and MGCP projects so I am a bit more conservative when discussing interop with those standards, however the SIP stack is very flexible and with the SIPaddheader() module you can do almost anything.
This sounds like an important project, have you considered engaging an outside consultant to support your project?
I assure you I do not want to lie to my carrier. These are the exact words that my carrier asked me so that my system does not have any problems when I connect to their sip trunk.
Is there more to connecting to a sip trunk on my end then typing the Peer and Client information and everything is set?
It seems like the sip trunk provider is being overly difficult.
Can you point me in the right direction for SIP Trunking interop I want to learn as much as I can.
Just typing the commands is the mechanical task, typing the correct commands requires knowledge of them and there are 100’s in modern Asterisk.
You didn’t mention the version you are running, the definitive guide to all the variables is the sample sip.conf in the Asterisk documentation.
use sipstation or flowroute
extremely asterisk friendly - save the money on the consultants and get better service