[OpenSIPS-Users] Opensips integration with asterisk

John Tuxies atuxnull at gmail.com
Wed Aug 21 08:47:32 EDT 2019


i have not had any progress with the integration.
My setup consists of:
-Opensips (2.4.2) with Control Panel and IP:192.168.1.250. Created through
the Control Panel users 2500-2509 and registered them to softphones and
they can talk to each other.
-2 Freepbx boxes, no configuration at the time, with IPs: 192.168.1.100 &
192.168.1.101 respectively
i am looking how to send the alter the configuration so that users in 2500
can access voicemail in the freepbxes and the rest of the services.
Also looking to register my ITSPs that require username/passwd/domain. Also
they forward numbers in DID mode and i have to assign them to particular
extensions. eg
DID_A to 2500
DID_B to 2501
DID_C to 2502
DID_D to 2503



i would appreciate all the help available. Once successful, i will document
everything and make it available to anyone looking to create something
similar.




On Mon, Aug 19, 2019 at 1:11 PM Alexey Kazantsev via Users <
users at lists.opensips.org> wrote:

> Hi John!
>
> >i am trying for some time now to integrate Opensips with Asterisk,
> > but without success. I have seen the links to the Opensips blog for
> > Asterisk integration, but it is outdated both for Opensips and Opensips.
>
> Well, what confused you in this tutorial? It seems to be what you need:
> https://www.opensips.org/Documentation/Tutorials#toc18
>
> >What i am trying to achieve is a box running Opensips with control panel
> >and another box with Asterisk. The reason for that is to enhance the users
> >with services such as IVR, Voicemail, email to voicemail, faxing,....etc
>
> >Up to now i managed to create users in Opensips and register on that.
> >Also they are able to make calls between them.
>
> You've written that you don't see such calls on the Asterisk.
> It means that you don't route such calls from OpenSIPS to Asterisk.
> Check this.
>
> >The numbering plan is 30XX and the port on the system is 5060.
> >Then i have another box with Asterisk that has the port as 55060
> >and the numbering plan is 30XX
>
> Well.. OK, let it be so.
>
> >and every time a user is created in Opensip's CP
> >then i create the same user in Asterisk,
> >eg Opensips 3000(port 5060) and Asterisk 3000(port 55060).
>
> But why?! You don't need this. Create SIP accounts either in OpenSIPS,
> or in Asterisk.
>
> >Then on the Asterisk box i made the following:
> >Created a trunk to Opensips
> >
> >[Opensips]
> >type=peer
> >host=192.168.1.113
> >context=from-opensips
> >insecure=port,invite
> >disallow=all
> >allow=alaw, g729, g722, ulaw
> >deny= 0.0.0.0/0.0.0.0
> >permit= 192.168.1.113/255.255.255.255
> >
> >
> >The problem is that i cannot see the call in Asterisk's terminal when 2
> users call each other.
>
> As I already wrote, it means that the call does not leave OpenSIPS.
> I guess you don't route it to Asterisk with smth like this:
>
>    ...
>    t_relay(x.x.x.x);   # Asterisk's IP
>    ...
>
> >Also , i have a couple of ITSPs in Asterisk that require username/passwd
> and thet have a FQDN.
>
> If you'd like to use OpenSIPS as the front-end, you'd better connect to
> ITSPs also from OpenSIPS.
> In case of SIP-registration,
> use UAC_REGISTRANT
> https://opensips.org/html/docs/modules/3.0.x/uac_registrant.html module.
> In case of SIP trunks just be able to receive SIP traffic from them and
> control access to your OpenSIPS
> public IP via iptables or PERMISSIONS module
> https://opensips.org/html/docs/modules/3.0.x/permissions.html.
>
> >While in Asterisk registered the user can access the first ITSP with the
> following prefixes 0 and 1 respectively.
> >Is there any way to allow the Opensips registered users dial 0 or 1 as
> prefix and place outgoing calls through ITSP 0 or 1, please?
>
> OpenSIPS is _extremely_ flexible.
> This could be achieved in many ways, starting from hardcoding in the
> script (in case of static configuration
> without need of changing it often) and ending with appropriate modules
> using, such as
> DROUTING https://opensips.org/html/docs/modules/3.0.x/drouting.html
>
>
> -----------------------------------------------
> BR, Alexey
> http://alexeyka.zantsev.com/
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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