And $ru is OK while sending to wrong (initial) IP? Did you try to set/check
$du too?
2015-05-15 11:30 GMT+03:00 Igor Potjevlesch <igor.potjevle...@gmail.com>:
> Hello,
>
>
>
> I experienced a strange issue with some of VoIP accounts.
>
>
>
> When the INVITE comes into MANAGE_FAILURE, after timeout, the config
> identifies, with "dialplan", the right Asterisk instance that should handle
> the call for voicemail.
>
>
>
> This part is okay, and results in a new INVITE with the Request-URI formed
> with the right domain (eg. sip:<NUMBER>@asterisk3). Then, the request
> goes to RELAY. Here is the issue: sometimes, the request is forwarded to
> the IP of the UA (the one initially contacted) instead of the IP of
> Asterisk.
>
>
>
> I can't figure out the difference between a succeeded call and a failed
> one.
>
>
>
> If someone has an idea. Here is the config that handles the VoiceMail:
>
>
>
> failure_route[MANAGE_FAILURE] {
>
> […]
>
> if (isflagset(24)) {
>
> $avp(s:inv_timeout) = "5";
>
> t_set_fr($avp(s:inv_timeout)*1000);
>
> if
> (avp_db_load("$to/username","$avp(s:vm_uri)/usr_vm")) {
>
> resetflag(24);
>
> avp_pushto("$ruri","$avp(s:vm_uri)");
>
> # Dynamic routing
>
> if
> (avp_db_load("$ruri/username","$avp(s:client)/usr_fai")) {
>
> if
> (dp_translate("2","$avp(s:client)/$avp(s:dest)") == 1) {
>
> $ru = "sip:" +
> $rU + "@" + $avp(s:dest);
>
> } else {
>
> # Load default
> voicemail
>
> $avp(s:client)
> = "DEFAULT_VM";
>
>
> dp_translate("2","$avp(s:client)/$avp(s:dest)");
>
> $ru = "sip:" +
> $rU + "@" + $avp(s:dest);
>
> };
>
> } else {
>
> # Load default voicemail
>
> $avp(s:client) =
> "DEFAULT_VM";
>
>
> dp_translate("2","$avp(s:client)/$avp(s:dest)");
>
> $ru = "sip:" + $rU + "@" +
> $avp(s:dest);
>
> }
>
> } else {
>
> xlog("L_WARN","time=[$Tf] call id=[$ci]
> call seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's
> method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message
> id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], User have no mail
> box\n");
>
> exit;
>
> };
>
>
>
> prefix("710");
>
> xlog("L_WARN","time=[$Tf] call id=[$ci] call
> seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's
> method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message
> id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], failure route to
> Voice Mail\n");
>
> route(RELAY);
>
> exit;
>
> }
>
>
>
> Regards,
>
>
>
> Igor.
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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>
>
--
Savolainen Dmitri
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