Discussion:
[Freeswitch-users] SIP Call-ID
Jed Stafford
2008-06-07 05:31:07 UTC
Permalink
I'm working to get FreeSwitch working behind a SIP load balancer. Using the
sip-ip param in a profile I'm able to make the outbound calls appear to come
from the load balancer IP address. Debugging the sip messages, this all
appears to be working correctly.

However the load balancer determines which machine to send the calls too
based on the SIP Call-ID. We want FreeSwitch in the middle of the call for
billing and other purposes. But i'd like FreeSwitch to use the same SIP
Call-ID that was sent to it when it initiates the next leg of the call. The
hash on the load balancer only cares about everything proceeding the @ sign
in the sip call-id. Is there a function in the dialplan to copy the source
call-id and use it for the destination leg?

Hopefully I've explained this well enough, any ideas would be helpful.

Regards,

-Jed
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080606/8ee0dd3f/attachment.html
Anthony Minessale
2008-06-07 15:34:00 UTC
Permalink
We didn't currently support that but I added a patch to trunk that should
let you do it.

sip_outgoing_call_id variable set on a new channel should let you choose
your own call id.

on the a leg ${sip_call_id} could be passed to the b leg one of 2 ways.

using the "export" app on the A leg: one call before the bridge

<action application="export" data="sip_outgoing_call_id=${sip_call_id}"/>
<action application="bridge" data="sofia/default/foo at bar.com"/>

using the originate syntax in the bridge.

<action application="bridge"
data="{sip_outgoing_call_id=${sip_call_id}}sofia/default/foo at bar.com"/>
Post by Jed Stafford
I'm working to get FreeSwitch working behind a SIP load balancer. Using the
sip-ip param in a profile I'm able to make the outbound calls appear to come
from the load balancer IP address. Debugging the sip messages, this all
appears to be working correctly.
However the load balancer determines which machine to send the calls too
based on the SIP Call-ID. We want FreeSwitch in the middle of the call for
billing and other purposes. But i'd like FreeSwitch to use the same SIP
Call-ID that was sent to it when it initiates the next leg of the call. The
in the sip call-id. Is there a function in the dialplan to copy the source
call-id and use it for the destination leg?
Hopefully I've explained this well enough, any ideas would be helpful.
Regards,
-Jed
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080607/d9a35f18/attachment.html
Loading...