hrm...it's also seems to be that if my lua script looks like
session:execute(bridge, sofia/gateway/XXX/0X)
session:execute(bridge, sofia/gateway//XXX)
if the first bridge fails, the session is immediately hungup, even if
hangup_after_bridge is set to false...is this the intended behavior?
grr...continue_on_fail...ignore my ignorance ;)
but it would still be nice getting a response back from the session:execute
bridge
--matt
On Thu, May 21, 2009 at 11:09 PM, Matthew Fong mattdf...@gmail.com wrote:
hrm...it's also seems to be that if my lua script looks like
would you like to try this?
bridge_hangup_cause = session:getVariable(bridge_hangup_cause) or
session:getVariable(originate_disposition);
if (bridge_hangup_cause == NORMAL_TEMPORARY_FAILURE or
bridge_hangup_cause == NO_ROUTE_DESTINATION or bridge_hangup_cause
== CALL_REJECTED)
2009/5/20 Michael Collins m...@freeswitch.org:
Lars,
Thanks for pointing this out. I will update the wiki. The new way to check
voicemail is to dial 4000 and then enter your extension.
What version is that on?
I just followed
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and
Thanks guys for a detailed reply specially pete.
On Tue, May 19, 2009 at 7:31 PM, Peter P GMX prometheus...@gmx.net wrote:
Thanks for this overwiev.
One question: How does this compare to Cepstral TTS?
Best regards
Peter
p...@privateconnect.com schrieb:
I've spent the last 2-3 months
Hello,
FS by design is B2BUA, and it cannot route INVITEs and other SIP methods. It
can however, bridge a-leg to b-leg with or w/o media and doing plenty other
cool stuff much better than commercial projects. I suggest joining us on irc
to detail your setup so we can help you.
Regards,
Ognjen
This is also interesting for me, as I love freeswitch, and maintaining a
single platform is easier, than handling various different ones.
In the past years I did a couple of projects with OpenSER /openSIPS.
These projects comprised:
* registrar for the SIP user agents
* handle invite
1) update to lastest trunk (you are at least 1000 revisions behind)
2) disable the presence debug in sofia.conf
3) enable sip trace instead sofia profile internal siptrace on
4) reproduce your problem.
Make sure you include more of the log from before the hangup happened.
The one you posted here
On May 22, 2009, at 12:47 AM, Jim Burke wrote:
Hey Brian,
Will have a look at ZRTP :)
Not sure I understand your comments regarding its all over once
receiving the 415 from the B party. Is'nt that what parm
continue_on_fail does? The fact that it sends the invite back out
sorta proves
You have old configs with the old method in it if you had a
previous install it won't overwrite the configs with new ones.
/b
On May 22, 2009, at 2:31 AM, Gavin Henry wrote:
What version is that on?
I just followed
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and registered
2009/5/22 Brian West br...@freeswitch.org:
You have old configs with the old method in it if you had a previous
install it won't overwrite the configs with new ones.
Ah, will re-install
___
Freeswitch-users mailing list
What problems will a Windows user have when updating with Tortoise SVN?
-Original Message-
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org
On Fri, May 22, 2009 at 7:58 AM, mszla...@aol.com wrote:
What problems will a Windows user have when updating with Tortoise SVN?
I haven't had a chance to test it out but what I would do is update and then
rebuild solution and see how it goes. Let us know if you run into any
issues.
-MC
Windows should have to clean the solution and rebuild
/b
On May 22, 2009, at 10:36 AM, Michael Collins wrote:
On Fri, May 22, 2009 at 7:58 AM, mszla...@aol.com wrote:
What problems will a Windows user have when updating with Tortoise
SVN?
I haven't had a chance to test it out but what I
Dasbus
On Thu, May 21, 2009 at 23:26, Diego Viola diego.vi...@gmail.com wrote:
Hey guys,
I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
cool names for my VoIP company, if you know some please tell me :)
Diego
___
VoiceCLOUD
CLOUDvoice
GlobalVoice
VoiceUP
voicEVERYthing
VoicEnterprise
S/IP (Services over IP)
GlobalSIP
VoiPLATFORM
Best regards,
-E
Gpro.ws
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
I say bkw_
On May 22, 2009, at 10:45 AM, SP wrote:
Dasbus
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Neospeech does have the best voices.
I looked at Neospeech months ago and talked to a rep.
Then he quoted me a price per port of over $1000.00. Looks like they really
have done some big price adjustments.
-Original Message-
From: Saeed Ahmad saeedahmad1...@gmail.com
To:
Just to followup on Cepstral. I used Cepstral for an Asterisk project a while back. So my information may be dated. I had not considered the for this project based on quality of voices.Pros:- Cheaper than other solutions ($50/port, $30/voice)- Well documented- Good Support- Supports MRCP- Has ASP
how about InterTalk or InterMedia?
-nandy
===
LanVox Systems
Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA: +1-360-8122281
http://sites.google.com/site/lanvoxphils
On Fri, May 22, 2009 at 11:52 PM, Brian West
Voila itself is a good name. ;)
VoilaVoIP
On May 22, 2009, at 12:26 PM, Diego Viola wrote:
Hey guys,
I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
cool names for my VoIP company, if you know some please tell me :)
Diego
How about larynxvoip
/b
On May 22, 2009, at 7:10 PM, dujinfang wrote:
Voila itself is a good name. ;)
VoilaVoIP
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
22 matches
Mail list logo
