Asterisk as B2BUA between OpenSIPS and SBN

Asterisk Asterisk Support
1

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:

[trunk-toip]
type=peer
insecure=very
context=from-toip
host=[SBC IP]

[voip-gw]
type=peer
insecure=very
context=from-voip
host=[OpenSIPS IP]
transport=udp

and on extensions.conf a simple:

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

[from-voip]
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

2

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.

3

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…

4

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.

5

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