Hi Bogdan,<div><br></div><div>Thanks for your reply.</div><div><br></div><div><br></div><div>Your script is very useful for calling between two opensips servers which I have tested.<br><br>However, I don't know how to configure on CUCM 7.0 which I am using.<br>
<br>At the moment, CUCM 7.0 is using Web Config via the Web Browser. <br><br>Can you let me know how to configure on CUCM 7.0?<br><br>I will appreciate very much if you give some instructions for configuring SIP Trunk on CUCM7.0</div>
<div><br></div><div><br></div><div>Thanks<br>Kind regards,</div><div>Steven,<br><br><div class="gmail_quote">On 10 January 2011 19:33, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi Steven,<br>
<br>
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.<br>
<br>
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:<br>
<br>
# all numbers starting with 55 are to be sent to CUCM<br>
if ($rU =~ "^55[0-9]+$") {<br>
# replace the domain part of RURI to point to CUCM<br>
rewritehostport("CUCM_IP:CUCM_PORT");<br>
# route the call out based on RURI<br>
route(1);<br>
}<br>
<br>
<br>
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.<div class="im"><br>
<br>
Regards,<br>
Bogdan<br>
<br>
steven chew wrote:<br>
</div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="im">
Hi Bogdan,<br>
<br>
Thank you very much for your reply.<br>
<br>
I have an Opensips Server and a Cisco Unified Communication Manager (CUCM).<br>
<br>
If I want to make calls from Opensips Server to CUCM and CUCM to Opensips Server.<br>
<br>
For example:<br>
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.<br>
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.<br>
<br>
Can you give some instructions how to configure the above scenario for dialing extension numbers?<br>
<br>
Thanks<br>
Steven, <br></div><div><div></div><div class="h5">
On 6 January 2011 21:31, Bogdan-Andrei Iancu <<a href="mailto:bogdan@voice-system.ro" target="_blank">bogdan@voice-system.ro</a> <mailto:<a href="mailto:bogdan@voice-system.ro" target="_blank">bogdan@voice-system.ro</a>>> wrote:<br>
<br>
Hi Steven,<br>
<br>
If you use the opensips default script, your opensips will accept<br>
calls from any other external SIP entities (call targeting a local<br>
opensips subscriber).<br>
<br>
If you want to configure your opensips to accept foreign calls<br>
only form a specific IP address, you can use the permission<br>
module, with address table to implement IP-based authentication.<br>
<br>
Best regards,<br>
Bogdan<br>
<br>
steven chew wrote:<br>
<br>
Hi everyone,<br>
<br>
I am a newbie with SIP-Trunk in OpenSips.<br>
I have a Cisco Communication Unified Manager and a OpenSips<br>
Server running in two different Virtual Machines.<br>
<br>
I would like to have a SIP trunk in between them "Cisco<br>
Communication Unified Manager and OpenSips Server".<br>
Therefore, I can make a call from OpenSips Server's SIP<br>
Clients to Cisco IP Phone.<br>
What should I need to add into opensips.cfg configuration file?<br>
<br>
I hope you can give some simple examples how to do it.<br>
I look forward to hearing from your advise asap.<br>
<br>
Thanks<br>
Regards,<br>
-Steven.<br>
<br>
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-- Bogdan-Andrei Iancu<br>
OpenSIPS Event - expo, conf, social, bootcamp<br>
2 - 4 February 2011, ITExpo, Miami, USA<br></div>
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</div></blockquote><div><div></div><div class="h5">
<br>
<br>
-- <br>
Bogdan-Andrei Iancu<br>
OpenSIPS Event - expo, conf, social, bootcamp<br>
2 - 4 February 2011, ITExpo, Miami, USA<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
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