[OpenSIPS-Users] How to implement a SIP Trunk in between two SIP servers.
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Tue Jan 11 09:25:03 CET 2011
Hi Steven,
sorry, but I know nothing on CUCM :(
Regards,
Bogdan
steven chew wrote:
> Hi Bogdan,
>
> Thanks for your reply.
>
>
> Your script is very useful for calling between two opensips servers
> which I have tested.
>
> However, I don't know how to configure on CUCM 7.0 which I am using.
>
> At the moment, CUCM 7.0 is using Web Config via the Web Browser.
>
> Can you let me know how to configure on CUCM 7.0?
>
> I will appreciate very much if you give some instructions
> for configuring SIP Trunk on CUCM7.0
>
>
> Thanks
> Kind regards,
> Steven,
>
> On 10 January 2011 19:33, Bogdan-Andrei Iancu <bogdan at voice-system.ro
> <mailto:bogdan at voice-system.ro>> wrote:
>
> Hi Steven,
>
> To do that, you need to add in opensips some routing to 1)
> recognize the numbers that needs to be sent to CUCM and 2)route
> that calls to CUCM.
>
> For script logic it sounds like : if you receive a new call
> (initial INVITE) for your local domain, check the URI and divert.
> If you look at the default config file, there is comment "#
> requests for my domain" -> from that point further you have only
> initial INVITEs for your local domain, so you can add after:
>
> # all numbers starting with 55 are to be sent to CUCM
> if ($rU =~ "^55[0-9]+$") {
> # replace the domain part of RURI to point to CUCM
> rewritehostport("CUCM_IP:CUCM_PORT");
> # route the call out based on RURI
> route(1);
> }
>
>
> For the other way around, you have to put a similar logic in CUCM,
> like to divert all calls starting with "12" to opensips - and
> replace the domain on RURI with the IP/domain of opensips.
>
>
> Regards,
> Bogdan
>
> steven chew wrote:
>
> Hi Bogdan,
>
> Thank you very much for your reply.
>
> I have an Opensips Server and a Cisco Unified Communication
> Manager (CUCM).
>
> If I want to make calls from Opensips Server to CUCM and CUCM
> to Opensips Server.
>
> For example:
> 1) If I dial an extension number "5566" from a SIP Phone
> "12345" under Opensips Server, it will try to call to a Cisco
> IP Phone "5566" from CUCM through a SIP Trunk.
> 2) If I dial an extension number "12345" from a Cisco IP Phone
> "5566" under CUCM, it will try to call to a SIP Phone "12345"
> under Opensips Server through a SIP Trunk.
>
> Can you give some instructions how to configure the above
> scenario for dialing extension numbers?
>
> Thanks
> Steven,
> On 6 January 2011 21:31, Bogdan-Andrei Iancu
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>
> <mailto:bogdan at voice-system.ro
> <mailto:bogdan at voice-system.ro>>> wrote:
>
> Hi Steven,
>
> If you use the opensips default script, your opensips will
> accept
> calls from any other external SIP entities (call targeting
> a local
> opensips subscriber).
>
> If you want to configure your opensips to accept foreign calls
> only form a specific IP address, you can use the permission
> module, with address table to implement IP-based
> authentication.
>
> Best regards,
> Bogdan
>
> steven chew wrote:
>
> Hi everyone,
>
> I am a newbie with SIP-Trunk in OpenSips.
> I have a Cisco Communication Unified Manager and a OpenSips
> Server running in two different Virtual Machines.
>
> I would like to have a SIP trunk in between them "Cisco
> Communication Unified Manager and OpenSips Server".
> Therefore, I can make a call from OpenSips Server's SIP
> Clients to Cisco IP Phone.
> What should I need to add into opensips.cfg
> configuration file?
>
> I hope you can give some simple examples how to do it.
> I look forward to hearing from your advise asap.
>
> Thanks
> Regards,
> -Steven.
>
>
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> -- Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami, USA
> www.voice-system.ro <http://www.voice-system.ro>
> <http://www.voice-system.ro>
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> --
> Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami, USA
> www.voice-system.ro <http://www.voice-system.ro>
>
>
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--
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami, USA
www.voice-system.ro
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