[OpenSIPS-Users] Route to media-server, but reply negative

Flavio Goncalves flavio at asteriskguide.com
Mon Oct 15 15:57:22 CEST 2012


Remco

Use hangup(3) and Asterisk will send a 404

Excerpt from
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause.

ISUP Cause value SIP response

1 unallocated number 404 Not Found
2 no route to network 404 Not found
3 no route to destination 404 Not found
16 normal call clearing --- (*)
17 user busy 486 Busy here
18 no user responding 408 Request Timeout
19 no answer from the user 480 Temporarily unavailable
20 subscriber absent 480 Temporarily unavailable
21 call rejected 403 Forbidden (+)
22 number changed (w/o diagnostic) 410 Gone
22 number changed (w/ diagnostic) 301 Moved Permanently
23 redirection to new destination 410 Gone
26 non-selected user clearing 404 Not Found (=)
27 destination out of order 502 Bad Gateway
28 address incomplete 484 Address incomplete
29 facility rejected 501 Not implemented
31 normal unspecified 480 Temporarily unavailable

Flavio E. Goncalves
SipPulse Routng and Billing Solutions for SIP.




2012/10/14 Remco . <remconl87 at gmail.com>

> Thanks Max. That does the trick for the Asterisk part. However, calls are
> now returned with 603-declined. Anyone on how to make opensips wait for the
> message to complete, and then return 404?
>
> On Sat, Oct 13, 2012 at 11:06 PM, Max Mühlbronner <mm at 42com.com> wrote:
>
>> Hi,****
>>
>> ** **
>>
>> regarding asterisk as media-server, you could use the “noanswer” option
>> for playback(). Then it will signal audio via progress messages but will
>> not answer (200 OK) the call.****
>>
>> ** **
>>
>> Best Regards****
>>
>> ** **
>>
>> Max M.****
>>
>> ** **
>>
>> *Von:* users-bounces at lists.opensips.org [mailto:
>> users-bounces at lists.opensips.org] *Im Auftrag von *Remco .
>> *Gesendet:* Samstag, 13. Oktober 2012 22:52
>> *An:* OpenSIPS users mailling list
>> *Betreff:* [OpenSIPS-Users] Route to media-server, but reply negative****
>>
>> ** **
>>
>> Hi all,
>>
>> I have the following in the failure_route, for invalid destinations:
>>
>>                 if(t_check_status("404")) {
>>                         # Dialed phone number does not exist
>>
>>                         # Cancel call billing
>>                         resetflag(1);
>>
>>                         # Start announcement
>>                         seturi("sip:AN_invalidnumber@[ip of
>> mediaserver]:5060");
>>                         t_relay();
>>
>>                         #t_reply("404", "Not found");
>>                         exit;
>>                 }
>>
>> When a 404 reply is received from upstream carrier(s) I would like to
>> play an announcement to let the user know they made a mistake in the phone
>> number.
>> On the other hand, I would like those calls to show up as '404' in my
>> statistics. Ideally the announcement should be played in the early media
>> (don't know if that's possible with Asterisk as a media server?).
>> The announcement works, however returns 200-OK. If I uncomment the
>> 't_reply', the call is ended to soon without allowing the announcement to
>> be played.
>>
>> Does anyone how to solve this? I tried branching but I cannot get it to
>> work.
>>
>> Thanks,
>> Remco.****
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20121015/52491278/attachment.htm>


More information about the Users mailing list