Go ahead and get a SIP trace with debug log of the call to that endpoint and put it on pastebin.freeswitch.org.

-MC

On Wed, Jul 10, 2013 at 3:08 AM, Ashwin Jain <ashwinrkjain@gmail.com> wrote:

Hi,

I am trying to connect asterisk to freeswitch. 

For this, I have added a contact in sip_profiles/users.xml, and asterisk is connecting FS as a sip contact. When I am trying to dial to asterisk contact, asterisk is not getting any SIP packets.

Following is the registration information of asterisk contact:

Registrations:

=================================================================================================

Call-ID:     4e46179c713c03f42d1a6dfe3b104777@127.0.0.1

User:       102@freeswitch.abcd.co

Contact:     "user" <sip:102@182.172.122.40>

Agent:       Asterisk PBX 1.6.2.12

Status:     Registered(UDP)(unknown) EXP(2013-07-10 03:09:02) EXPSECS(128)

Host:       freeswitch

IP:         182.172.122.40

Port:       14638

Auth-User:   unknown

Auth-Realm: freeswitch.abcd.co

MWI-Account: 102@freeswitch.abcd.co

Total items returned: 1

=====================================================================

--

Thanks and Regards,
Ashwin Jain

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--
Michael S Collins
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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