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-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@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 
<mouli...@gmail.com<mailto:mouli...@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 
<mouli...@gmail.com<mailto:mouli...@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:

http://pastebin.com/fr8m9gr7

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:

http://pastebin.com/y4tXXrnF

Any update would be appreciated.

Thanks in advance.

Regards,
Chandra.





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