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Re: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?


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  • From: Stephen Kingham <>
  • To:
  • Subject: Re: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?
  • Date: Mon, 11 Jul 2005 09:53:03 +1000

Hi Dinesh

Sounds like you have made some good progress.

Dinesh wrote:

Hi Stephen,

Well today, I managed to get something to work.

I replaced one nic on the SER machine, and using the xten lite phone behind
ser. I was able to call between my asterisk server and the ccm using sip.
I already have ccm and Asterisk talking. But the problem happens, I think
because of proxy redirects in ser. Public to Public IP is fine. But what I
wanted to do is Public to Private.

Yes you are hitting the NAT problem. So you can do it in SER but you have to add the NAT helper stuff and I have not done that myself as I am in the fortunate position of having enough public address space.

Since my asterisk is on a public ip and a private ip and can talk to my ccm.
Just want to ask the sip.edu community, if we use the asterisk as a bridge
between ser and the ccm? Asterisk can call ccm, and ccm can call asterisk.
Now can ser call ccm going through asterisk?


So you are really using Asterisk to support the NAT and if it works then why not. Then use SER for the routing etc in front of Asterisk should work.

Stephen, some pointers on the firewall mapping and the placement of the SER
would be useful:)


This would be a good section under SECURITY in the Cookbook. Here is a bit of a start (I think this sort of issue needs a diagram):

SIP Server listens on port 5060 UDP and TCP.

Open up Hi Ports coming from the outside going to port 5060 on the SIP Servers (for SIP).

You will also need to open up Hi UDP from the outside to the Asterisk and
from Asterisk to the outside (for RTP).


Regards,

Dinesh.



-----Original Message-----
From: Dinesh [mailto:] Sent: Wednesday, June 29, 2005 9:11 PM To:
Subject: RE: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?

Hi Stephen,

I had 2 nic cards on the SER machine. One internal and One external. I was
on the internal address, talking to the SER machine's external address for
registering my xten phone. Obviously when I tried to register it took my
nat public address. The reason why I did this, is because my ccm is on a
internal address.

Now I am thinking of just removing one nic card and trying to contact my
Asterisk Server with the SIP external IP address. As I have asterisk server
on a public ip. I will try it later if this works to the same with call
manager or do some intresting firewalling natting for 5060 for my call
manager.

To answer your question, I tried to call extension(9804 or 65869804) which
is behind the call manager from the SER SIP phone (xten lite).



On the call manager make sure the SIP Trunk on the CCM has a Calling Search Space (CSS) that is allowed to make calls to extensions, ie the CCS has the necessary Partitions added. - Please excuse the Cisco Call Manager speak ;-).


I know what exactly you mean, when I was interfacing the cisco call manager
and asterisk, I had to worry about that. Spend a quite a day trying to
figure this out.
I will follow your other email also and get back to you shortly. I will
also try the SER with one NIC instead of 2.

I think it would be great with the screen shots. In Singapore there is
quite number of ccm installations, specially in the gov sector. I think if
you guys made a screenshots, more people will be able to join the sip.edu
project.



Regards,

Dinesh.


-----Original Message-----
From: Stephen Kingham [mailto:] Sent: Tuesday, June 28, 2005 3:52 PM
To:

Subject: Re: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?

Hi

Can I presume you have the CCM calling the SER SIP Phone?


Dinesh wrote:


Hi Stephen,

I tried to configure this. But I am having some problem to talk to the ser
(from ccm4.0). What is the string you used to talk or dial a extension

from

ser to ccm 4.0? I assumed you used sip trunk between the ccm4.0 and ser.





The string is the dialed digits. CCM ip address 192.168.63.242
Dialed digits 3575

In SER
rewritehost("192.168.63.242");
then route call with
t_relay();

On the call manager make sure the SIP Trunk on the CCM has a Calling Search Space (CSS) that is allowed to make calls to extensions, ie the CCS has the necessary Partitions added. - Please excuse the Cisco Call Manager speak ;-).

You might need to look at an ngrep trace (try ngrep -p -q -W byline port 5060)


Good luck and let me know how you go as I can probably help a bit more that these comments.

I guess we should document this with some screen shots for the Cookbook.

Stephen


Regards,

Dinesh.









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Stephen Kingham, MIT, BSc, E&C Cert
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