OT: Disabling 3pcc in CallManager 7.1
Hello everyone, I've got a SIP trunk up to a Cisco CallManager 7.1 system. I've been able to get it to work by enabling 3pcc in FreeSWITCH but I'd like to disable 3pcc and be able to have a standard INVITE w/SDP, 183, 200, conversation. Does anyone have any clue how to configure CallManager appropriately? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
Nope that thing is EVIL... I haven't the foggiest clue how to do that we have a jira open on this issue. /b On May 11, 2010, at 12:31 PM, Kristian Kielhofner wrote:
Hello everyone,
I've got a SIP trunk up to a Cisco CallManager 7.1 system. I've been able to get it to work by enabling 3pcc in FreeSWITCH but I'd like to disable 3pcc and be able to have a standard INVITE w/SDP, 183, 200, conversation. Does anyone have any clue how to configure CallManager appropriately?
Thanks!
--
Someone from Voiceops pointed me in the right direction. I'm waiting to confirm but this might be the ticket: "Media Termination Point Required" is checked "on" in the SIP trunk configuration window. That should force an SDP in the initial INVITE... On Tue, May 11, 2010 at 1:41 PM, Brian West <brian@freeswitch.org> wrote:
Nope that thing is EVIL... I haven't the foggiest clue how to do that we have a jira open on this issue.
/b
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
We need to wiki that if at all possible once you confirm that it works. /b On May 11, 2010, at 1:29 PM, Kristian Kielhofner wrote:
Someone from Voiceops pointed me in the right direction. I'm waiting to confirm but this might be the ticket:
"Media Termination Point Required" is checked "on" in the SIP trunk configuration window.
That should force an SDP in the initial INVITE...
KK, Thanks, as usual, for sharing the knowledge that you so diligently earn through hard work. Please keep up the good work. :) Also, we are booking speakers for ClueCon... we'd love to have you back to talk about something interesting. Let us know! Thanks, MC On Tue, May 11, 2010 at 11:29 AM, Kristian Kielhofner <kris@kriskinc.com>wrote:
Someone from Voiceops pointed me in the right direction. I'm waiting to confirm but this might be the ticket:
"Media Termination Point Required" is checked "on" in the SIP trunk configuration window.
That should force an SDP in the initial INVITE...
On Tue, May 11, 2010 at 1:41 PM, Brian West <brian@freeswitch.org> wrote:
Nope that thing is EVIL... I haven't the foggiest clue how to do that we have a jira open on this issue.
/b
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
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Haha thanks MC but the real credit goes to Mark Holloway on Voiceops for pointing me in the right direction. I have tested this with CUCM 7.1 and confirmed it works. Where should this be documented? I'll see if I can come up with anything "interesting" for ClueCon... On Tue, May 11, 2010 at 3:01 PM, Michael Collins <msc@freeswitch.org> wrote:
KK, Thanks, as usual, for sharing the knowledge that you so diligently earn through hard work. Please keep up the good work. :)
Also, we are booking speakers for ClueCon... we'd love to have you back to talk about something interesting. Let us know! Thanks, MC
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
On Tue, May 11, 2010 at 1:22 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
Haha thanks MC but the real credit goes to Mark Holloway on Voiceops for pointing me in the right direction.
I have tested this with CUCM 7.1 and confirmed it works. Where should this be documented?
Since you're so gracious: http://wiki.freeswitch.org/wiki/Cisco_Call_Manager
I'll see if I can come up with anything "interesting" for ClueCon...
Sounds good. If you've done anything new with Recqual that would be fine,
or some other interesting VoIP app... -MC
MTP required on the Cisco SIP trunk will turn on "early media negotiation". This means it will send an SDP with its initial Invite, rather than waiting on the other side to send SDP. Hope this helps. Ken -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Tuesday, May 11, 2010 1:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1 Someone from Voiceops pointed me in the right direction. I'm waiting to confirm but this might be the ticket: "Media Termination Point Required" is checked "on" in the SIP trunk configuration window. That should force an SDP in the initial INVITE... On Tue, May 11, 2010 at 1:41 PM, Brian West <brian@freeswitch.org> wrote:
Nope that thing is EVIL... I haven't the foggiest clue how to do that we have a jira open on this issue.
/b
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
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Brian West
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Ken Fulmer
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Kristian Kielhofner
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Michael Collins