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Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy

Hi all, I am looking for phones witch support POE, with a good relation between quality and price to work with asterisk. I just see the Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave you the best results in a productivity enviroment? Thanks in advance. VoipCrazy. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the "Recording" button on this phone. The extension I configured on this phone has the values "Recording on demand", an the voicemail enabled. I am using FreePBX to manage my PBX. How should I configure the "Function

Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with QSIG configuration? Any clue will be wellcomed. Thanks Voipcrazy -------------- next part

Hello Voipcrazy, It's funny you should mention that - we've just released (as in today) a FREE version of our OrderlyStats service for call centre and queue monitoring and management. OrderlyStats features realtime (synchronous/message-based) display of all queue, agent and caller events so you can see what's happening in your call centre as it happens. The control panel feature

Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser ..... Is it necesary run a SER server on this enviroment? Any clue will be welcomed. Thanks in advance. VoipCrazy

Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the

Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

Dear list, I am trying to configure the nagios plugin called check_sip. I just read the README file included with the plugin. I follow the readme steps to configure the plugin, without success. In the nagios web interface I can see (No output!) In the status information column. If I run the chech_sip plugin from a linux console, I get /usr/local/nagios/libexec# ./check_sip -u

Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you tested? Has anyone experience about "vhost", "vserver", or something similar to run

Hello all, I am trying to set up asterisk and hylafax to send and receibe fax. The machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port). My problem is that , when I send a Fax from the PSTN to this machine, the asterisk or diva or hylafax, does not detect this call as a fax and asterisk answer that call like a voice call. How sould I configure the software modem (iaxmodem) to use

Dear all, I have got a PRI line with E1 20 channels, my question is: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 If this is my zaptel config for an E1 PRI line, Which would be the config for a reduced PRI line for 15 channels? and for 20 channels? Thanks in advance. VoIPcrazy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I want to analyze the asterisk logs files, looking for all kind of errors, ?Anyboby knows any asterisk logs analyzer? Thanks all, Voipcrazy

Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any "messages index file" or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the

Hello, I need to connect asterisk 1.2.16, with a Contect Center software that works with TAPI. As I know, asterisk doesn't support TAPI directly, if needs a tirth party software. I just reading about asttapi and Activa TAPI. does anyone test this software? have you using asterisk againts a TAPI compatible software? Witch TAPI software do you test? Thanks in advance. VoipCrazy