Callerid(name) between 2 asterisks


Have: 2 same asterisk’s servers (centos 7) v.16.3.0 + realtime + pjsip driver.
CallWay: Server 1(User’s Endpoint => Trunk’s endpoint) => server 2(Trunk’s endpoint => User’s endpoint).

Problem: server 1 user’s Endpoint dont sees called name, only called number from server 2 user’s Endpoint. That called name absent in sip trace. However there is called name at place server 1(Trunk’s endpoint) => server 2(Trunk’s endpoint) . It just vanished at 180 and 200 to caller user.

Question: How can I do that caller user could see called user’s name? That works only within 1 server.

server to server (problem)
within 1 server (no problem)

Please, help.

Please provide the logs from Asterisk, not a third party tool.

You are doing something wrong. Withouth details of your configuration, it is not possible to work out what, although one thought is that you have 2843 on both Asterisks, with type=friend, and it is being used in preference to the “trunk”, and also is explicitly setting the caller ID.

I dont have same endpoints on both asterisks.

There is one question: why asterisk dont transmit user part of “from”, “pai” and “rpid” SIPheaders to caller’s shoulder when call provides between 2 asterisks

How can I post any logs, if I can’t upload files (cause new user), and if I cant link it from my googledrive (cause your forumsystem find it as a spam) ?

Please provide SIP traces as provided by Asterisk. You didn’t even say which IP address was which in your previous traces, so wrongly associated an INVITE as being the B leg one. I’m not going to waste more time by trying to match INVITEs to calls in your current log.

My expectation is that Asterisk will pass the whole of the name and user part through unless you do something to overwrite one of them.

Calltrace: => => => user2(none in trace)

In 180 ringing and 200ok from to you can see user part of “pai” and “rpid” SIPheaders (many ticks)
P-Asserted-Identity: “… … 138” sip1011170@voip-rozru
Remote-Party-ID: “… … 138” sip:1011170@voip-rozru;party=called;privacy=off;screen=no

In 180 ringing and 200 ok from to there is no user part in same SIPHeaders
P-Asserted-Identity: sip:1011170voip-srvru
Remote-Party-ID: sip:1011170@voip-srvru;party=called;privacy=off;screen=no

Now its clear?

Im ready to give you log from asterisk cli, but I cant, cause take a hiden mod for this case (cause url for
doodle drive).

Responses don’t carry caller ID!!

Ok, I have found this endpoint’s item.
Set trust_id_inbound to yes. With this item endpoint dont overwrite 2nd leg callerid to himself, even endpoint’s callerid = NULL.

You can close this topic.

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