Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take
more than a couple of minutes to compile, install, and configure the package. Maybe some things
need to get adjusted as the author has abandoned the project (at least there is no longer a
project web
Hi people
I'm just mailing to see what people are using for CTI solutions with
asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e:
http://camrivox.com/products/flexor-cti-salesforce/
We've used this for a few clients.
On Fri, Jan 10, 2014 at 6:33 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi people
I'm just mailing to see what people are using for CTI solutions with
asterisk. Aslos, has anyone managed to integrate
Hi everybody,
I'am developping an interface to manage asterisk follow me module.I want
to do this in realtime and I didn't find the structure of the asterisk
follow me module.
I'am working with asterisk 1.8.5 and Postgresql 9.2 on CentOS 6.2
Can any body help me !
Thanks.
--
- Original Message -
http://camrivox.com/products/flexor-cti-salesforce/
We've used this for a few clients.
How were your experiences with it? I have a customer that will want this type
of integration in the near future, and would love to hear how installation,
operation, and
On connection to an incoming call via PSTN where
asterisk [11.7.0] is Dialing an internal extension
on answering the call there is about 6-7 seconds before
audio is heard on either side.
When looking at the CLI traces when I answer the incoming call that
asterisk extensions were dialing, I see
Hello Folks;
I have an Asterisk server
Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on
2013-12-27 18:47:44 UTC
No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk.
Is there an API out there that anyone knows of that I can pass commands,
etc to Asterisk? Creating
No major issues. They're always very responsive. I'd get a demo from
them for the client and make sure that the feature set is a match. But
I always say that with 3rd party apps.
On Fri, Jan 10, 2014 at 10:39 AM, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
2014/1/9 Shaun Ruffell sruff...@digium.com
On Thu, Jan 09, 2014 at 06:01:34PM +0100, Olivier wrote:
Hi,
On a Asterisk 1.8.12 system working OK for months (100k calls proceed),
users are complaining for bad audio.
My setup is:
PSTN --E1/PRI --- Asterisk --- E1/PRI--- Siemens HiPath
On Fri, Jan 10, 2014 at 07:48:21PM +0100, Olivier wrote:
With a single span directly connected to PSTN I'm still getting timing
slips (140 slips/hour).
Would you agree to qualify this rate as excessive ?
Yes, this is excessive.
Given these figures, may I also exclude an hardware failure
2014/1/10 Shaun Ruffell sruff...@digium.com
On Fri, Jan 10, 2014 at 07:48:21PM +0100, Olivier wrote:
With a single span directly connected to PSTN I'm still getting timing
slips (140 slips/hour).
Would you agree to qualify this rate as excessive ?
Yes, this is excessive.
Given these
On Fri, Jan 10, 2014 at 08:34:57PM +0100, Olivier wrote:
2014/1/10 Shaun Ruffell sruff...@digium.com
You've configured the card to recover timing from the provider?
I'm not sure but I don't think so as I've just configured the card with:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
Hello,
Anyone know good quality text to speach engine for building IVRs for
asterisk. Open-source will be nice, but I wont mind paying for thing really
good.
Regards,
-Jai
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Search google for Asterisk Manager Interface.
For the most part, if you have raw Asterisk installed then that's what you get
and have to build what you want on top of it or hire a developer to do it.
Date: Fri, 10 Jan 2014 12:12:47 -0500
From: szilvertho...@gmail.com
To:
Luminvox is one.. There are others out there..
Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448
From: jpra...@gmail.com
Date: Fri, 10 Jan 2014 12:16:43 -0800
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Text to Speech Engine
Hello,
Anyone know
Actually, scratch that.. Luminvox is not text to speech it's speech recognition
software. Got this mixed up and turned around :-) Anyhow, see the link I posted
earlier, it's got some good info to get you started.
From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Fri, 10 Jan 2014
On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com wrote:
On connection to an incoming call via PSTN where
asterisk [11.7.0] is Dialing an internal extension
on answering the call there is about 6-7 seconds before
audio is heard on either side.
When looking at the CLI
Lumenvox is actually both... but hard to justify for TTS given all the
freebies...
On 01/10/2014 02:50 PM, Todd R. wrote:
Actually, scratch that.. Luminvox is not text to speech it's speech
recognition software. Got this mixed up and turned around :-) Anyhow,
see the link I posted earlier,
On 10/1/14 8:16 pm, Jai Rangi wrote:
Anyone know good quality text to speach engine for building IVRs for
asterisk. Open-source will be nice, but I wont mind paying for thing really
good.
We recently used Ivona for a fairly complex IVR project (multi-lingual,
including pronunciation of
http://translate.google.com/translate_tts?tl=enq=i always find google
translate works well
http://translate.google.com/translate_tts?tl=frq=je trouve toujours google
translate fonctionne bien
On Jan 10, 2014 3:17 PM, Jai Rangi jpra...@gmail.com wrote:
Hello,
Anyone know good quality text to
Thank you every one,
Yes google's translate is really good.
http://zaf.github.io/asterisk-googletts/
But I dont like the fact that have to go over the wire every time. Looking
for some thing to install on local server.
-Jai
On Fri, Jan 10, 2014 at 5:15 PM, Darryl Moore dar...@moores.ca
On 01/10/2014 04:01 PM, Matthew Jordan wrote:
On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com wrote:
On connection to an incoming call via PSTN where
asterisk [11.7.0] is Dialing an internal extension
on answering the call there is about 6-7 seconds before
audio is heard on
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