CUCM 10.x with Third Party PBX & IP-PBX

[sabir.bedi1]

Team,

Please guide me with best possible options to integrate Third Party PBX & IPPBX with CUCM 10.X

Alcatel/Nortel of any other Vendor .

Need complete Network Diagram call flow and integration possibilities 

[Dennis Mink]

it depends, if these alcatel/nortel are IP based, run a SIP trunk between them, if they are not, then possibly qsig.

if you want a more detailed answer, provide model numbers please

[sabir.bedi1]

Hi Dennis Mink,

Thank you .I will try to find out model number soon 

But Tell me one more thing - If customer wants to remove their existing IP-PBX(Third party) and wants everything to be on same platform thats Cisco(Customer has CUCM) ,How can we integrate legacy phones to cucm ?removing the IP-PBX from environment ?

i believe using QSIG we might need to manage legacy pbx as well as cucm   

[Commsouth Infocom]

Hi Sabir,

If the customer existing phones are SIP supported, then you can register as 3rd party SIP Phones in CUCM. If they are not SIP supported and if it is analog, you can do it via Cisco Voice Gateway.

Regards,

Prasana

[Dennis Mink]

cant really answer that unless you can tell me what sort of phones they are, if the phones support SIP, you can pobably register them on call manager as 3rd party sip phone, otherwise; not gonna work.

[Suresh Hudda]

The best way is to keep the existing epabx on and integrate it with cisco, as in Nortel we used to use digital phones (3902,.3903, 3904 etc) so most probably you have digital phones as well and these phones won't work with Cisco call manager.

And Nortel IP phones works on UNISTIM protocol so they also can't register to Cisco call manager.

So migrate users in phases to cisco and integrating Cisco call manager with existing systems will not add much cost.

You can integrate Nortel pbx using qsig as blow (if you VTRK server at Nortel side then use SIP)..

Cisco cucm ---- Cisco mgcp voice gateway ----pri over qsig ---- DDP2 Nortel pri card ----- Nortel pbx (11c, 81c or cs1k).

Alcatel IP phones support sip so they might register to Cisco call manager but you won't get support from Cisco for these end points if some features won't work (conference, call transfer etc)

Suresh

[sabir.bedi1]

Suresh Hudda & Dennis Mink 

Need Urgent help 

Can you tell me how many participants can be on call simultanoeusly using Cisco 3905 Phone ?

I cant see any document that says it can handle more than 3? Also some says it doesnt  even use conference resources to make a conference more than 3 participants 

[SaibhuwanSudina70631]

Hello Mr.Suresh,

Please suggest me what yo do in the below scenario.

we have few grand stream phones and few Cisco IP phones.

Now, there is also a CUCM, which i am managing and....a third party unknown IP-PBX ( i don't know the model or name ), which is located remotely.

now grand stream phones are registered on the IP-PBX, where it is located remotely. and Cisco IP-Phones are registered by me on the CUCM which is present it the same office.

now the scenario is like....

I have created a SIP trunk on CUCM with the destination IP of that remote unknown IP-PBX. SO CUCM was successfully communicating with remote unknown IP-PBX.

I don't know what they have configured, but grand stream phones are able to communicate with Cisco Ip Phones with an extension of 8888 ( eg : 88882001 ). but when i tried to make a call from Cisco IP phones to Grand stream phones...., the call is not getting placed.

Now, the requirement of customer is like, call processing should be happening between, 

1) Cisco phones - Cisco phone ( working fine )

2) Grand stream phones - Grand stream Phones ( working fine )

3) Grand stream phone - cisco phones ( working fine ) ( with extension of 8888 )

4) Cisco phones - Grand stream phone ( solution required ) ( please suggest what to do )

Thanks & Regards

Sai Bhuwan

[janarna]

Hi Saib-- what was the fix you did on your issue with Grandstream IPBX to Cisco IPBX?

[b.winter]

I don't know, if he fixed the problem at all, but I suspect it was a call routing issue.
I would suggest you first talk with the grand stream admin how numbers should be called and then you have to format the calling and called numbers that way.
If it is not working, troubleshooting 1-0-1: check the logs of CUCM in RTMT. As it is a SIP trunk, you already see the messages in the Real-Time data section in RTMT.