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Rangel [mailto:cran...@globaltelesourcing.com]
Enviado el: jueves, 26 de junio de 2014 01:27 p.m.
Para: mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls
Hi Daniel
Thank you so much for your response. Here
-users@lists.sip-router.org
Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
Hi,
a call trace could help you to understand why your server is not receiving
response (which one actually ?) form the Cisco.
Is that because the Cisco didn't receive your SIP message? Or is it be
Hi,
a call trace could help you to understand why your server is not
receiving response (which one actually ?) form the Cisco.
Is that because the Cisco didn't receive your SIP message? Or is it
because Cisco replied but the response didn't reach your server ?
Try to make a sip trace in order to u
Hello,
can you gran the SIP trace on kamailio server for such case?
You can use ngrep, like:
ngrep -d any -qt -W byline port 5060
and send the output to the mailing list. You can replace any sensitive
information (e.g., ip address) before sending to mailing list.
The typical call drop after
Hello
I have successfully (I believe) implemented Kamailio 4.1.4 integration with
Freepbx 5.2.11 taking as a guide Daniel's tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
I just did not create the voicemail tables because voice mail is handled by
Freepbx.
