[SR-Users] Kamailio stopped relaying SIP text message from asterisk

Karsten Horsmann khorsmann at gmail.com
Mon Nov 25 10:02:33 CET 2019


Hi Adesh,

sngrep is an great debugging tool for that purpose.

You can use the siptrace module for recording your sip-messages in a
database.

route[MESSAGE] {
    if (is_method("MESSAGE")) {
        xlog("L_INFO", "[MESSAGE] this is an MSG from <$fU> to <$rU>\n");
        sip_trace();
    }
    return;
}

Example playaround config from me:
https://github.com/khorsmann/kamailio-keepalived/blob/master/roles/kamailio/templates/etc/kamailio/kamailio.cfg.j2#L596

Cheers
Karsten

Am Mo., 25. Nov. 2019 um 08:37 Uhr schrieb Adesh Pandey <
adesh.pandey at myoperator.co>:

> Hi,
> I have been working on kamailio with following scenario:
>
> 1. Webrtc SIP Users register to Kamailio
> 2. Users initiate outgoing calls to some other numbers
> 3. Kamailio dispatches outgoing calls to Asterisk
> 4. Asterisk answers the call and then send a text message to the caller
> using SendText, which was working fine a few days back but all of sudden it
> has stopped working
> 5. At Asterisk end it is showing that message was delivered successfully
> but when I check SIP user logs, there is no message event.
>
> Please review my config file at following URL:
> https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4
>
> Developers, I also want to know if there is any way to log *text message*
> in kamailio server.
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Mit freundlichen Grüßen
*Karsten Horsmann*
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