Re: [asterisk-users] Microsoft CRM Integration

2013-07-16 Thread Steven Howes
On 16 Jul 2013, at 04:10, Klaverstyn, David C wrote: I’m hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Hi, You've not said what you're trying to integrate... Creating tasks for calls, contact

Re: [asterisk-users] Microsoft CRM Integration

2013-07-16 Thread A J Stiles
On Tuesday 16 July 2013, Klaverstyn, David C wrote: Hi All, I'm hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Well, that's a bit of a vague request. If you just want to add the facility to

[asterisk-users] Extra Sound Packages

2013-07-16 Thread jg
Maybe this is a stupid question. Are the files in Extra Sound Packages related to any product or are they just supplemental material? I searched the source files for some of the file names and didn't find any reference. jg --

Re: [asterisk-users] ignore 183 session progress in parallel call scenarios

2013-07-16 Thread Steve Davies
I am sure I submitted the following alternative behaviour to the bug-tracker in the past, but cannot find any reference to it. Here is the patch I use to IMHO improve this behaviour. In case it is not officially uploaded, I will state here that this code is disclaimed and unencumbered as if

Re: [asterisk-users] Microsoft CRM Integration

2013-07-16 Thread David Wessell
http://www.camrivox.com/products/flexor-cti-dynamics-crm/ -- Ringfree Communications David Wessell 828-575-0030 x101 From: Steven Howes steve-li...@geekinter.netmailto:steve-li...@geekinter.net Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] Voice analytics

2013-07-16 Thread Julian Lyndon-Smith
Does anyone know of a realtime voice analytic engine that works with asterisk 11+ ? We want to be able to listen on the conversation for key words in order to ensure compliance . The plan is to show these keywords onscreen, and remove them once the agent has covered the compliance issues. This

[asterisk-users] FLAC script to convert from wav to FLAC and also with other 3 to 4 formats

2013-07-16 Thread Gopalakrishnan N
Hi, Below link is the script which i found while surfing, this script basically converts your voice file to flac format, where the file is reduced to 50%. http://legroom.net/files/software/convtoflac.sh The quality is really good, I tested. this... In large production environment this script

Re: [asterisk-users] ignore 183 session progress in parallel call scenarios

2013-07-16 Thread Hristo Trendev
Thanks Steve! I too believe that this is indeed much better handling of 183 replies in a parallel call. After testing for several hours today I actually wen't a bit further (see ASTERISK-22082) and proposing to ignore the 183 altogether as far as parallel calls are concerned. In my case I don't

[asterisk-users] Help with decyphering DND status

2013-07-16 Thread James B. Byrne
Arch x86_64 OS CentOS-6.4 (freepbx) Asterisk 11.4 FreePBX 2.11.0.4 Snom870 with FW-8.7.4.8 What I am attempting to do is to set a different background colour for the BLF vkeys when a station is set to DND. This is supposedly accomplished through this setting in the phones provisioning file:

Re: [asterisk-users] Mailing a fax with mutt does not succeed

2013-07-16 Thread Daniel - Asterisk
Hello everyone, I'd changed the server and mutt started working, but I'll test your advices and wil let you lnow ass soon as I can. Thank you! Elder On Mon, Jun 24, 2013 at 7:38 AM, Larry Moore lmo...@omninet.net.au wrote: On 22/06/2013 2:17 PM, Steve Edwards wrote: On Sat, 22 Jun 2013,

Re: [asterisk-users] PoE module

2013-07-16 Thread Niles Ingalls
Here's a cheap solution for PoE piggybacked over your existing network. http://www.amazon.com/gp/product/B0002R6X9S On Jul 14, 2013, at 3:12 PM, bilal ghayyad wrote: Hello; We have a cisco switches but they are not PoE and we need only to have PoE device so the cables come for it first to

[asterisk-users] SIP timers

2013-07-16 Thread Deka, Rajib IN MAA SL
Hello List, I tried to change the following parameters in sip.conf file, but looks like it cannot be changed, Defaut values: ;t1min=100 ;timert1=500 ;timerb=32000 I have changed to: ;t1min=100 timert1=100 timerb=6400 Sometime I can see too many retransmission of BYE to some of the UAs