similar to: Call Center software opensource or commercial

Hello, We use and develop the astGUIclient suite. It is Open-source(as in GPL) and offers Inbound and Outbound call center functions with reports, ACD, monitoring, recording and very basic IVR scripts. Complex IVR functions need to be custom programmed within Asterisk but that is not really that hard. It works across multiple Asterisk servers and we are using it currently at 5 locations including

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

Hello, I am moving topic from asterisk-dev list to asterisk-users list. Did anyone succeed receive incoming calls in h323 and orient them to right context based on "host" identification? To summarise, I have quintum Gateway sending call to Asterisk box, and I would like to use asterisk as a protocol converter h323 --> sip. in h323.conf, I have [quintum_gw1] type=user

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

On Fri, Jul 20, 2001 at 11:27:16AM +0500, Asif M. Baloch wrote: > Hi guys, > > ICE cast resembles shoutcast, thats for sure and its good that its open > source. But, it has too many probs. I ran it on a dual p3 800 with a T3 comm We are running icecast on linux 2.2.19 serving more than 600gb each months and it has been rock stable so fare. Right now our uptime is around 3 months.

I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one : Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> ) exten =>

Hi, I intend to change the cdr_mysql-field "uniqueid", which seems not to be used so far, to an (not unique) indexed field and use it later for my own hints and infos. I don't have very much traffic so far, and I wonder, if there will appear problems when asterisk is under high load (100 simultanious calls) and the log table contains 1.000.000 log lines. This would mean, that

On 10/17/19 4:08 PM, Johnny Hughes wrote: > On 9/24/19 2:41 PM, Frank Cox wrote: >> Without wanting to sound too pushy, I'm wondering if there is any update on the status of Mate now that Centos 8 has been released? >> >> I would love to jump on C8 and start playing with it, but the lack of Mate is kind of a showstopper for me at the moment. >> > > Is gnome3

Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have

You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. ________________________________ From:

I need for asterisk to call me at a predetermined number once a day at a predetermined time and once connected to me make 5-10 simultanious calls to a DID filling all available channels. What is the best way to do this? Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091015/b97f2d1e/attachment.htm

I appreciate any recomendation of a simple CallingCard Application and resources of users manual. __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

Just good old monitor with no mixing onto the scsi drive. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk - sound quality-critical! Hi, how do

Hello, I'm trying to sum up current options for doing small (up to 20 agents) inbound-only CC. I've found: astguiclient, maybe there are some other CC solutions? And on the other side: witch is better to have 20 PC w/ softphones or one T1 channel bank and normal phones with hands-free sets. (whitch set whould you recomend) ? kd, -- Krzysztof Drewicz Affordable 2/4 span E1/T1

Hello Rubyists, I''m in the position of being given the job to design and build a mission-critical web facing application for a small but growing enterprise. It is to be used by customers as needed, numbering in the tens up to the hundreds. It will collect operational data, particularly in a table around 50 columns wide, and potentially millions of rows deep, most fields being numerical.

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I

YANS: Yet Another Silly Idea The discussion about tags seems to me to be getting a little silly. >From my 'newbie' perspective on tag formats, it seems to me that all tags are arbitrary. I mean, sure, you could add a 'your player should support this tag in this format', but really, it comes down to consistency on the Encoder's (person that encoded the file) part in how to

On Wed, 9 Feb 2005, Stefan Gofferje wrote: > I wanted to try out app_icd but... > > root@k-tanco:/opt/app_icd> make > ===> Compile: /opt/app_icd/app_icd.c (app_icd.o) > app_icd.c: In function `app_icd__log_events': > app_icd.c:2104: error: structure has no member named `cid' > app_icd.c:2104: error: structure has no member named `cid' > make: ***

Hi Im using Icecast2 and SAM2 to broadcast a 56kb 24khz Stereo 2 Channel 6.Kb MP3 stream on a 512k ASDL connection. I know that the number of listeners will be LIMITED to 3 or so - is there any way I can actually calculate this ? I passed my broadcast link to a friend yesterday and they told me that they recieved a 'Server Full' error, is there something in the config that I need to

Hi Thomas, I don't know if you like top or bottom quoting. That seems to be a big-little endian thing ;-) So, I will top quote and inline quote. Please see my comments inline, below. Kind regards, Stephan On 02/19/2015 04:18 PM, "Thomas B. R?cker" wrote: > Hi, > > On 02/19/2015 03:07 PM, Stephan Leemburg wrote: >> Hello Icecast-dev, >> >> I am new to