Aterisk IPBX system and CUCM 9.1 trunking help needed

[Ajith Nayak]

Dear Team,

We are using FreePBX 12.0.32 with core is Asterisk version 13.0.0, hardware now is VMWare ESX 5.0 virtual machine.

IP Phone is AudioCodes 310HD and Soft phone is any sip client, we use CouterPath X-lite.

1.SIP trunk between Asterik and CUCM 9.1 is it possible and supported? if yes will cisco TAC can help incase of any issue or toubleshooting?

2.Can my CUCM registerd IP phones will be able to make calls to Asterisk registerd IP phones vice versa?

3.H/ow will be the dial plan,What changes we have to make in the IPBX system?

4.Please share any document which will help me to achive it .or any example live scenario etc.

Regards,

Ajith

[Vivek Batra]

1. Yes, you can create a SIP Trunk from CUCM to Asterisk. It's available under Device --> Trunk in CUCM. Not sure but I think TAC should support this as it's standard SIP based on RFC 3261.

2. From CUCM to Asterisk using SIP trunk, it will be possible to route calls from end points registered with CUCM.

3. In CUCM, you need to make appropriate Route Pattern, Route Group/List and SIP trunk configuration.

4. Please check the following link for SIP trunk setup in CUCM;

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_1_1/ccmcfg/CUCM_BK_A34970C5_00_admin-guide-91/CUCM_BK_A34970C5_00_admin-guide-91_chapter_01001000.html#CUCM_RF_SBB2D11C_00

[Ajith Nayak]

Hi Vivek,

Thanks for your reply

From CUCM i can say I will create required trunking I am comfrtble from that part

Any specific ports should be opened on firewall side? to achive this?

Is there anything which we have to configure from Astrisk IPBX side? any manupulation?  or any dial plan? I am not sure on that part.

rgds,

ajith

[Vivek Batra · Accepted solution]

If firewall is in between CUCM and Asterisk, I suppose only default SIP port 5060 needs to open in firewall. I assume RTP will pass directly b/w the end points otherwise RTP ports also to be adjusted in firewall.

In Asterisk, yes we also need to create SIP trunk similar to CUCM. Note that this will be peer-to-peer trunk (no registration). Dial plan will also be required in Asterisk to route the call to respective extension of Asterisk and  viceversa.

Whether you need any digit manipulation or not in Asterisk depends on your number schema. If you keep simple dial plan with unique numbering pattern, I suppose you can avoid any type of digit manipulation in Asterisk (and in CUCM as well :).

[Ajith Nayak]

Thank Vivek Batra

IT was very helpful :) :) !!!

Rgds,

Ajith