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nitesh.bansal at gmail...
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 Posted: Wed Apr 13, 2016 3:05 pm Post subject: [asterisk-users] Using Asterisk to route call via an outboun |
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Hello,
I want to use Asterisk to use Kamailio as an outbound proxy for routing calls to remote SIP end points, one option could be to use a default peer, but in my case, my outbound proxy can change
based on the remote end point, so this option doesn't work.
And another problem is that I don't know how to configure Asterisk to prepare the Request-URI
based on the remote end point and not based on the outbound proxy address?
What is the best way to do it?
Thanks,
Nitesh |
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jcolp at digium.com
Guest
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| Posted: Wed Apr 13, 2016 3:09 pm Post subject: [asterisk-users] Using Asterisk to route call via an outboun |
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Nitesh Bansal wrote:
| Quote: | Hello,
I want to use Asterisk to use Kamailio as an outbound proxy for routing
calls to remote SIP end points, one option could be to use a default
peer, but in my case, my outbound proxy can change
based on the remote end point, so this option doesn't work.
And another problem is that I don't know how to configure Asterisk to
prepare the Request-URI
based on the remote end point and not based on the outbound proxy address?
What is the best way to do it?
|
You'll have to be specific with the channel driver in use. Speaking from
chan_pjsip it does not have a mechanism to set the outbound proxy on a
per-call basis, it's strictly controlled by the endpoint. You'd need
multiple or construct endpoints dynamically (for example using the ARI
push configuration). As for not rewriting the request URI you need to
use loose routing by specifying ;lr in the outbound proxy URI.
Example:
sip:example.com;lr
If used in a configuration file:
sip:example.com\;lr
The '\' is so the configuration parser does not treat it as a comment.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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nitesh.bansal at gmail...
Guest
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| Posted: Wed Apr 13, 2016 3:11 pm Post subject: [asterisk-users] Using Asterisk to route call via an outboun |
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I'm using chan_sip, I experimented with adding a 'Route' header in the originate command and used the Dial command like 'SIP/peer/exten', but problemis that Request-URI isn't populated correctly.
I'm using Asterisk 13.
Thanks,
Nitesh
On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
| Quote: | Nitesh Bansal wrote:
| Quote: | Hello,
I want to use Asterisk to use Kamailio as an outbound proxy for routing
calls to remote SIP end points, one option could be to use a default
peer, but in my case, my outbound proxy can change
based on the remote end point, so this option doesn't work.
And another problem is that I don't know how to configure Asterisk to
prepare the Request-URI
based on the remote end point and not based on the outbound proxy address?
What is the best way to do it?
|
You'll have to be specific with the channel driver in use. Speaking from chan_pjsip it does not have a mechanism to set the outbound proxy on a per-call basis, it's strictly controlled by the endpoint. You'd need multiple or construct endpoints dynamically (for example using the ARI push configuration). As for not rewriting the request URI you need to use loose routing by specifying ;lr in the outbound proxy URI.
Example:
sip:example.com;lr
If used in a configuration file:
sip:example.com\;lr
The '\' is so the configuration parser does not treat it as a comment.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
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nitesh.bansal at gmail...
Guest
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| Posted: Wed Apr 13, 2016 3:22 pm Post subject: [asterisk-users] Using Asterisk to route call via an outboun |
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I'm also using ARI to dynamically select the Asterisk peer and remote end point, concern is how to use the same peer configurationwith different end points and have asterisk populate the request-uri correctly?
Nitesh
On Wed, Apr 13, 2016 at 10:11 PM, Nitesh Bansal <nitesh.bansal@gmail.com (nitesh.bansal@gmail.com)> wrote:
| Quote: | I'm using chan_sip, I experimented with adding a 'Route' header in the originate command and used the Dial command like 'SIP/peer/exten', but problemis that Request-URI isn't populated correctly.
I'm using Asterisk 13.
Thanks,
Nitesh
On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
| Quote: | Nitesh Bansal wrote:
| Quote: | Hello,
I want to use Asterisk to use Kamailio as an outbound proxy for routing
calls to remote SIP end points, one option could be to use a default
peer, but in my case, my outbound proxy can change
based on the remote end point, so this option doesn't work.
And another problem is that I don't know how to configure Asterisk to
prepare the Request-URI
based on the remote end point and not based on the outbound proxy address?
What is the best way to do it?
|
You'll have to be specific with the channel driver in use. Speaking from chan_pjsip it does not have a mechanism to set the outbound proxy on a per-call basis, it's strictly controlled by the endpoint. You'd need multiple or construct endpoints dynamically (for example using the ARI push configuration). As for not rewriting the request URI you need to use loose routing by specifying ;lr in the outbound proxy URI.
Example:
sip:example.com;lr
If used in a configuration file:
sip:example.com\;lr
The '\' is so the configuration parser does not treat it as a comment.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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jcolp at digium.com
Guest
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| Posted: Wed Apr 13, 2016 3:32 pm Post subject: [asterisk-users] Using Asterisk to route call via an outboun |
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Nitesh Bansal wrote:
| Quote: | I'm also using ARI to dynamically select the Asterisk peer and remote
end point, concern is how to use the same peer configuration
with different end points and have asterisk populate the request-uri
correctly?
|
I can only speak for chan_pjsip I'm afraid, but Olle has provided
information about chan_sip on your -dev cross post.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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