[OpenSIPS-Users] Unable to route registered phone to registrant trunk
Bogdan-Andrei Iancu
bogdan at opensips.org
Wed Jun 3 17:04:18 CEST 2015
Hi Fabrizio,
First of all, be sure your script execution gets to the do_routing()
part. Use the script_trace() function to trace the execution of your script:
http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc43
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.06.2015 18:29, Fabrizio Pappolla, Brainy Forge S.r.l. wrote:
> Hi.
> I'm using opensips-1.10.
> I have successful configured the uac_registrant module to my VoIP
> carrier, so my OpenSIPS is registered; I've also correctly configured
> some user: 1000 and 1001. I can register to OpenSIPS with this
> extension and they can call each other. If I try to call external
> number I get "SIP 404 - NOT FOUND" error.
>
> How can I make external call? I've tried to configure drouting module,
> without any success.
>
> /opensipsctl dr show/
> *dr gateways*
> +----+------+------+-----------------+-------+------------+-------+------------+-------------+
> | id | gwid | type | address | strip | pri_prefix | attrs |
> probe_mode | description |
> +----+------+------+-----------------+-------+------------+-------+------------+-------------+
> | 2 | 1 | 0 | voip.eutelia.it | 0 | 0 | 1
> | 2 | eutelia |
> +----+------+------+-----------------+-------+------------+-------+------------+-------------+
> *dr groups*
> +----+----------+-----------------+---------+-------------+
> | id | username | domain | groupid | description |
> +----+----------+-----------------+---------+-------------+
> | 1 | 1000 | voip.eutelia.it | 1 | |
> +----+----------+-----------------+---------+-------------+
> *dr carriers*
> +----+-----------+--------+-------+-------+-------------+
> | id | carrierid | gwlist | flags | attrs | description |
> +----+-----------+--------+-------+-------+-------------+
> | 1 | 1 | 1 | 0 | | EUTELIA |
> +----+-----------+--------+-------+-------+-------------+
> *dr rules*
> +--------+---------+--------+---------+----------+---------+--------+-------+-------------+
> | ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
> attrs | description |
> +--------+---------+--------+---------+----------+---------+--------+-------+-------------+
> | 1 | 1 | | | 0 | 1 | 1,#1
> | | |
> +--------+---------+--------+---------+----------+---------+--------+-------+-------------+
>
>
> /opensipsctl dr gw_status/
> ID:: 1 IP=voip.eutelia.it *Enabled=no*
>
> If i try to enable it, it goes down after few sec
>
> /opensipsctl dr carrier_status/
> ID:: 1 Enabled=yes
>
> Is the dynamic routing way correct? I need to forward all sip traffic
> from a user (that register to OpenSIPS) to its associated trunk ( by
> uac_registrant module ) and opposite.
> Regards.
>
>
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