SIP Trunking with asterisk

[LineMann]

Hi,
I want to interface my unify 0s4k with asterisk PBX application. Can anyone please guide me on this issue. I am getting :service unavailable: 503 " issue. Kindly suggest what should be the proper settings for the TDCSU to interface with asterisk PBX.
Thnx

 

[sbcsu]

Not sure but have a look at the 'docs' folder
faq965-7883

 

[Moriendi]

You don't say if the 503 is one direction or both. You need to trace it.

If you can't dial from 4K to Asterisk you need a wireshark trace to see if an invite is generated out from the STMI. If it is, config sounds pretty much OK but you need to check digits. If no invite is sent, it's 4K config for sure.

If you can't dial from Asterisk to 4K you need an RDS trace from the Assistant to check if the STMI generates a QSIG setup into the 4K.

Without a trace you are just guessing.

 

[thepfy]

503 could be a failure to negogiate a codec.

There is a guide in the Unify wiki but don't know how old it is.

[URL unfurl="true"]https://wiki.unify.com/images/c/c4/Asterisk_HiPath_4000.pdf[/url]