Re: [sipx-users] How to integrate SIPX 4.0 with Call manager 5.0

["Picher, Michael" <mpicher@xxxxxxxxxxxxxxxx>]

I just replied to somebody else on the list with seemingly the same problem…  I’ll respond here as well just for the sake of this thread.

 

If the traffic is coming from the IP address of the CM server, use the IP address in your gateway definition on the sipXecs side.  If the traffic is coming from a SIP domain, you should use the SIP domain in the gateway definition and make sure that your sipXecs server has the proper DNS records to resolve the domain name.

 

This really is much the same concept as connecting two sipXecs servers via gateways:  http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_two_sipX_PBXs

 

 

Mike

 

 

 

From: Emery ville [mailto:emeryvilleberkley@xxxxxxxxx]
Sent: Wednesday, July 01, 2009 7:00 PM
To: Picher, Michael
Cc: sipx-users@xxxxxxxxxxxxxxxxxxx
Subject: Re: [sipx-users] How to integrate SIPX 4.0 with Call manager 5.0

 

Hey Mike,

I am able to make calls one way from phones on sipXecs server to phones on call manager but cannot make calls in the other direction,
no calls from phones on call manager to phones on sipXecs server.

I made call manager as the unmanaged Gateway and gave the dial plan it works only one way please help me make it two way.

Thanks in advance,
Emery

On Tue, Jun 30, 2009 at 6:47 PM, Picher, Michael <mpicher@xxxxxxxxxxxxxxxx> wrote:

That’s not how I’d go about it…

 

I’d setup the Call Manager box as a gateway on the sipX box and then make a dial plan that looks for the extensions on the Call Manager box and routes them out that gateway.

 

Mike

 

From: sipx-users-bounces@xxxxxxxxxxxxxxxxxxx [mailto:sipx-users-bounces@xxxxxxxxxxxxxxxxxxx] On Behalf Of Emery ville
Sent: Tuesday, June 30, 2009 9:26 PM
To: sipx-users@xxxxxxxxxxxxxxxxxxx
Subject: [sipx-users] How to integrate SIPX 4.0 with Call manager 5.0

 

Hello friends,

 

I am having trouble integrating SIPX 4.0 with Call manager 5.0

 

I established a sip trunk between both but the sip trunk never is up, I am actually not sure if the sip trunk is up.

 

I cannot make any calls between the servers.

 

what I did :

Logged in to Https as superadmin and did the following:

 

1)

 

I hit the tab System==>servers==>sipxecs_server==>

and then in the configure options of left side menu

I checked the SIP truinking option.

 

 

2)

 

Devices==> SBC==> created internal sbc.

 

3)

 

Then the Devices tab on top

 

Devices==>Gateways==> right side drop down menu ==>sip trunk==>

 

configuration : name XXX ip ; address : ip add of call manager, route: sipxbridge-1

 

 ITSP account : ITSP server domain name : 10.x.x.194 (call manager ip address)

username : XXX

password : XXX

 

4) system==> dialplans==>Dial rule ==>

checked enabled

 at the bottom of the page the gateway

 

restarted all that is required.

 

I see that the sip is working good when I hit the tab System==>servers==>sipxecs_server==> and services.

 

Please help me, I thank you in advance.

 

Thanks,

Emery

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