I will know monday morning if Patch21 will do the job. I just made a 1
hour an 10 minute test call from my google voice number with no voice
drop. We'll see how it goes
BTW I know EXACTLY what you mean about laborious and lengthy QA. I have
to do it all the time for new voice applications and pr
You will most likely get better input from sipx-dev instead on sipx-users.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.84
I'm trying to load a custom FreeSWITCH module for use by the Media Services
in sipXecs. Steps taken so far are:
1. Installed the module to the mod directory @ /usr/local/freeswitch/mod
2. Added load statement to the modules.conf.xml file
3. Reboot the server
When using the FreeSWITCH CLI tool (/
I have no doubt that it will work with freeswitch when running through
the patch. Same thing with YaTE.
I don't think my FreeSWITCH settings would be a good thing to post on
the wiki because I essentially turned off all SIP security settings in
an effort to get it working. YaTE, on the other ha
Thanks for the update Ranga.
Josh, if you do apply patch21.zip and have this working with freeswitch on
pfsense, please let the list know. I would suggest using a separate posting.
Perhaps a write-up for the wiki so others can follow as well if you have the
time and get that far, with an example c
On Sat, Nov 14, 2009 at 1:00 AM, Josh Patten wrote:
> It also appears that YaTE is the same way. that one was a little easier to
> set up, but it's the same old song and dance: REFER trips it up every time.
>
> I really wish sipXbridge was stable for me. Even with patch20 I drop to one
> way audio
> software from vendors that can adhere to problems escalations and rapid
> software fixes as needed. The burdens for this in the open source world,
> due to the legal needs to always evolving e911 regulatory requirements, are
> pretty harsh.
I was hoping that the system was fairly standardized by
Hi,
I'm new to sipxecs. Currently I have done a proof of concept with
sipxecs and different phones for a customer. My main focus until now was
the Cisco voip world.
I noticed the same problem with my cisco vg 2811 and all SIP Phones
trying to do a pickup.
I'm using version 4.1, there I notice
Hi Tony,
I did understand you perfectly :-)
Having had a few hoirs sleep, what you want to do is called
"Presentation Restriction" which, as long as the carrier honours it,
will restrict the presentation of "your" caller ID to called parties.
To have it apply to all calls, I'd suggest the simple
Maybe others have had better luck... I certainly haven't.
It's a bit different with regular PBX's or breaking out analog lines in
a router like you're doing, in that they'll take in a PRI or Analog line
and leave the signaling analog (never converting to IP).
Mike
-Original Message-
Fro
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