Asterisk as B2BUA between OpenSIPS and SBN

Hi all,

i’ve build up an Asterisk 18.10 LTS to use as B2BUA agent between OpenSIPS and an SBC. Both are SIP peers so I’ve configured sip.conf as:

host=[SBC IP]

host=[OpenSIPS IP]

and on extensions.conf a simple:

exten => s,1,Dial(SIP/voip-gw/${EXTEN})

exten => s,1,Dial(SIP/trunk-toip/${EXTEN})

but doesn’t work and Asterisk, when an INVITE arrive, say to me:

chan_sip.c:26824 handle_request_invite: Call from ‘voip-gw’ ([IP]:5060) to extension ‘0[NUMBER]’ rejected because extension not found in context ‘from-voip’.

Any help really appreciated

Thanks, MP

That’s quite true. Only extension s exists in that context. You need to add something that matches the extension actually used,

This only exists in a deprecated, unsupported, driver, due for removal next year. Having said that:

This is a deprecated version of both insecure=invite and insecure=port. The former has no effect without a secret. Are you sure you really need the latter. You haven’t specified any codecs. This can cause problems. There is an open bug for allow=all, and even if you don’t trip that bug, you can create excessively long INVITE requests.

Thanks for your reply. I’m trying move to PJSIP stack but i’m pretty new to that and i’m trying to orientate myself… any hint? Any help? I just need a simple and basic B2BUA…

You seem to have a very simple, IP based, unauthenticated, connection on both sides. It is too simple for an example to be provided, but take a look at the provider trunk example in res_pjsip Configuration Examples - Asterisk Project - Asterisk Project Wiki and delete everything to do with registration, authentication, and setting the from user.

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