Subject: SIP in higher education
RE: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?
- From: "Dinesh" <>
- To: <>
- Subject: RE: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?
- Date: Mon, 11 Jul 2005 10:10:55 +0800
Thanks to all, specially Stephen for giving me information on how to get the
sip.edu working. I will be fine tuning the scripts, access-lists, ser and
For now, I haven't looked at dialing by email address, but I have installed
the mapping to phone numbers. For example,
ring my extension 6586 9804. This is for all IMCB users who have a valid
extension. I still have to do the other way, IMCB users calling the sip.edu
From: Stephen Kingham
Sent: Monday, July 11, 2005 7:53 AM
Subject: Re: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?
Sounds like you have made some good progress.
>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
>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
>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
>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).
Stephen Kingham, MIT, BSc, E&C Cert
Project Manager and Consulting Engineer
Telephone +61 2 6222 3575 (office)
+61 419 417 471 (mobile)
Voice and Video over IP
for The Australian Academic Research Network (AARNet)
- Re: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?, Stephen Kingham, 07/10/2005
- RE: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?, Dinesh, 07/10/2005
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