1. Asterisk should accept all calls from Kamailio - at least this is my setup.
2. Kamailio should accept all calls from Asterisk servers. Again - this is my setup.
3. All calls which are not from Asterisk servers are directed to some asterisk server.
4. All calls from any of Asterisk server are directed to their target.
This way I have User(100....500)<->Kamailio<->Asterisk, with usage of rtpproxy. For ACK/HANGUP - additional checking.


On Mon, Jun 16, 2014 at 7:18 PM, Waite, Hugh <hugh.waite@acision.com> wrote:

Hi,

 

I’m not a big Asterisk user so I may not be able to help much here.

To encourage a helpful response, please provide some information on what you are expecting to happen and not just what isn’t happening.

 

If you are expecting a second phone to ring, this is likely to be a signalling issue. You said that the clients are registered on Kamailio. Are they also registered on the Asterisk server? The second and third lines in your logs say “Subscriber Absent/Everyone is busy” which suggests they are not.

A network trace taken on the Kamailio or asterisk server will show if the REGISTER is being forwarded to Asterisk. Are there any errors logged when the REGISTER is processed by Asterisk which indicates a failure?

 

It looks like the system falls back to voicemail, so you are also expecting the Asterisk server to play media. This will be a separate media issue.

I have used rtpproxy (with the advertised address patch) in Amazon to bridge media between internet facing and private subnets in a VPC. This may or may not be necessary depending on whether the Asterisk server has a public IP address.

 

I found that I couldn’t use different advertised addresses depending on which direction the signalling was going on a single private IP address. I worked around this by allocating a second private ip address to the instance and used that in the ‘bridge’.

-A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15

 

Hope that helps you get a bit further.

 

Hugh

 

From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Chandramouli P
Sent: 16 June 2014 10:34
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Unable to make calls within the extensions

 

Hello,

I really don't understand why I am not getting any reply for my query. Is it the wrong mailing list for my question?

Can anybody confirm?

Thank you.

Regards,
Chandra.

 

On Fri, Jun 13, 2014 at 5:48 PM, Chandramouli P <mouli123@gmail.com> wrote:

Hello,

 

Can anybody please respond?

 

Any update would be appreciated.

 

Thank you.

 

Regards,

Chandra.

 

 

On Wed, Jun 11, 2014 at 12:55 PM, Chandramouli P <mouli123@gmail.com> wrote:

Hi,

I am new to Kamailio. I started RTPProxy using "rtpproxy -A 54.85.12.15 -F -l 10.0.0.122 -s udp:localhost:7722" command and see that my sip phones are registered with Kamailio. I am able to see using "kamctl ul show" command. But, I am unable to establish call between my registered extension through Asterisk using Kamailio. I see that calls are hitting my Asterisk server when I call from extension. I am not getting any audio and the other extension is also not at all ringing. I could not able to figure out where I am doing the mistake? I am not sure whether the mistake is in Kamailio or Asterisk.

In my Asterisk server, I am seeing the correct configuration using "odbcinst -q -d" and "isql -v MySQL-asterisk chandra test123456" and "odbc show (At CLI>)".

Please find the below environment:

Operating System: Ubuntu 14.04 Server (64-Bit)
Kamailio: 4.0
Asterisk: 11.10
Database: MySQL (UNIX ODBC)
Environment: Amazon EC2
Follwed Links:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
http://pastebin.com/VPpfErYn
Daniel's Patched RTPProxy Installation (Please note that My Asterisk and Kamailio servers are behind NAT in Amazon EC2)

Below is my configuration:

 

 

Note: I inserted the records in to the respective tables with "cmp" as context.

 

When I call from 100 to 500, I am not getting any sound and another extension is not ringing and getting the below messages at Asterisk CLI:

 

 

Any update would be appreciated.

 

Thanks in advance.

 

Regards,
Chandra.

 

 

 

 


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