similar to: Call restriction for particular extension

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

Can't I upload the 3PCC firmware ? available from the Cisco website? Actually it came with sip88xx.... firmware. Regards . On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies147 at gmail.com> wrote: > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

Hello all, I have the code below to simulate samples of certain size from a particular distribution (here,beta distribution) and compute some statistics for the samples. betasim2<-function(nsim,n,alpha,beta) { sim<-matrix(rbeta(nsim*n,alpha,beta),ncol=n) xmean<-apply(sim,1,mean) xvar<-apply(sim,1,var) xmedian<-apply(sim,1,median)

Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=

Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I" from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it

Hi List, Were upgrading our network switches and need to create multiple VLAN groups, but since our Squid Proxy (Transparent Proxy) Server should be accessible to all VLAN groups we need to setup a trunk grouping inside our Squid Proxy Box. Is anyone has a documentation or code on how to implement trunk grouping? Your thoughts will be highly appreciated. Regards, Malvin

Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -------------- next part

Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > Its really weird working with OpenSuse. I am not sure how others are using > with OpenSuse. Through Yast also I tried to install Asterisk package, it > didn't find. > > Now I am clueless to work with OpenSuse. > >

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Actually now I have the phones with SIP firmware. I will try with 3pcc firmware along with XML files. Or any idea if we have CUCM application can we change the firmware. am ready to buy the developer edition. Regards . On Mon, 5 Dec 2016, 3:34 p.m. Steve Davies, <davies147 at gmail.com> wrote: > I tried... repeatedly... I failed. I bought some 3PCC phones, and they > just worked.

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to upload woth TFTP due to some reason it's getting failed. Do I need to load 3pcc firmware or anyway to Configure from the phone itself or from the GUI? I have the SEPMAC.cnf.xml as well. Any suggestions would be appreciated. Regards .