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cucm 6.0 and sip analog gateway

Petr Simanov
Level 1
Level 1

hello all

i connect d-link analog gateway dvg-5008s to callmanager

each port of gateway as third-party sip device

but, i can register just one port, any one port, not all

where is trouble?

9 Replies 9

Petr Simanov
Level 1
Level 1

problem is solved

just connect gateway as third sip device (advance) and configure dn's on it's lines

JATINDER KUMAR
Level 1
Level 1

hi Petr...

do you have any idea how to integrate sip device to cucm 6.1. In my enviorment wee have a sip device (Third party) on which i have configured my voip account.it working fine if i am connecting analog phone to that deivce and dialing some number but now i want that device to integrate with call manager and i want to give this service on IP Phones can u plz help me in this..

hello Jatinder

i dont understand what kind of service u want give to sip device

hi Petr,

     Thanks for the quick reply, ok i will explain the infrastructure right now we have got Cisco call manager 6.1 running giving ip telephony + unity service (extra server) to about 100 users using an ISDN line to make all the calls but the problem with this setup is to make international calls this setup is very costly so we want to reduce the cost on this ... so we got a device which is actually a sip device (planet technologies) and i configured sip setting on this and it works fine when connecting a analog phone and this is very-2 cheap to make international calls call goes through internet now i want to use this sip device with my call manager so that when ever some one dials an international number it goes through SIP device..

hope u got the answer of your question. Please help if you have any idea

Jatin

Jatinder,

u just need to configure on callmanager a few things:

1) create sip trunk to you sip device

2) create route pattern and route calls that u need (international) to this new sip trunk

3) if u want, u can attach pattern to some css

best regards

hi petr,

i did the same thing but when dialing its giving fast busy signal.. dont know .. how can i check that when i am dialing international its going to my sip device.

hello,

check, that your phone attached to correct css, that contains your pattern

to test it, go to cisco unified serviceability on cucm, than tools, and Dialed Number Analizer

than Analisis and Analizer

Calling Party is number from what you calling

Dialed Digits, i think u understand what is it

than choose correct css and do analysis

in log u can see where your call go

to what device, isdn line or sip trunk

Thanks Petr.. Thanks an ton...will let u know

thanks agian

Leo Gonzales
Level 1
Level 1

Hello,

I hope I have the right discussion group.

I have a Cisco 2851 CUCM 6.0. I have 1-4port  voice FXO and 1-4port voice FXS in router. I am trying to add a device gateway to connect a regular pots line phone. I attempted to add a gateway template and used 8989 as the directory number. I deleted the template and attempted to add the device gateway under the device menu. However, now when I try to add the DN, the CUCM indicates "Add failed. Could not insert new row - duplicate value in a UNIQUE INDEX column".... How can I remove the DN 8989 from the database so that I can use it when I create the gateway?

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