search for: salama

Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel

Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? Why is this affecting the VoIP world so much and not the regular PSTN analog world? What does the PSTN industry have that they can handle such high volume of calls and there is "no" echo problem? Thanks, Daniel

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

I just installed a TE410P on a Debian Sarge system running kernel 2.6.11-1-686-smp. Zaptel and Asterisk seem to be working fine. However, I have a couple of problems with the TE410P and Zaptel. First, the TE410P is showing me red alarms on 2 of the 4 T1s. This board (the TE410P) was just moved from another machine running REL3 and all 4 T1s were working there. I don't know why only

Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Thanks, Daniel

I have an Perl AGI script that acts as an IVR for my Asterisk box. Basically, it simply plays audio files to the caller, collecting DTMF input and logging every DTMF input into a database table, simply to document every step or option selected by the caller. One thing is that in addition to playing audio files, it also, at some point, plays SayUnitTime command. This IVR has constantly

...that we use in house. We are interested in working with others on a version 2 skipping some of the mistakes of our first version and using a better model. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX virtualization Any alternate open-source solutions? On May 25, 2006, at 2:17 PM, Douglas Garstang wrote: > Yes, but it fast becomes a provisioning and managem...

...3800 (Wasif) > 8. Re: Need to Hire: PHP Programmer for PhoneCALL (olivier.taylor) > 9. Re: Anyone see this? (Matt) > 10. Re: SIP codec preference order ineffective (olivier.taylor) > 11. Re: Comedian Mail not deleting .txt file (Doug Lytle) > 12. Re: WRTG54GS Capacity (Daniel Salama) > 13. Re: DUNDi Not Able to Handle Complex FailoverSituations > (Stephen Davies) > 14. Re: sip to h323 gateway ... (Tigran Kocharyan) > 15. RE: Best $300 VoIP phone for asterisk? (Colin Anderson) > 16. RE: DUNDi Not Able to Handle ComplexFailoverSituations > (Dougla...

Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? Thanks, Daniel

I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the

Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not familiar with network configuration really, although I do understand most of the basics. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version:

Does anybody have experience with running ferret on dreamhost ? My app is running ok until I install the acts_as_ferret plugin, at which point I get "Rails application failed to start properly" errors. I''ve used script/console to confirm that I can require ''ferret'' and make a new Index object . Everything appears to be ok in that respect. Unfortunately

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

I'm setting up a demo for two asterisk machines. One will be a central Asterisk server which will handle everything already in VoIP (office-like functions plus agents functionality). The second Asterisk box will be used strictly as a VoIP gateway to the first server. The gateway server will have 4 T1s connected to it and what I was thinking on doing was the following: in

I'm noticing by watching the CLI that my inbound calls coming via T1s on a TE410P are using GSM codec. Why wouldn't it use ULAW as default? How can I make it use ULAW as default? Thanks, Daniel

I just came across http://www.voip-info.org/tiki-index.php? page=Asterisk%20TDMoE and seemed very interesting. It prompted me to question whether it would be more efficient to do TDMoE or IAX2. The application is very simple. I have two asterisk boxes. One is strictly a gateway to the PSTN using multiple T1s. The second is where all my extensions and dialplan are defined (the workhorse for

Can anyone provide more information on switch or point me to where I can find more about it? The only I've been able to find on the wiki is: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers and towards the bottom of (section Forwarding to another Asterisk): http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Some of the questions I have are: 1) If I have an