similar to: Query about Call Recording with Asterisk / Freeswitch in Cisco IPCC deployment

Hello, We are going to start development for a product based over Asterisk. According to you, which is the preferred language for AGIs / IVRs development in Asterisk. I got opinions that Perl is going to be replaced by PHP for all future developments. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif

Hello All We need to connect our client's offices located in Dubai and Pakistan. Suggest us some economical solution. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN: kashif__naeem at hotmail.com Gmail: meet.kashif at gmail.com Skype: kashif.naeem 302 Y Commercial Area,

Hello All We have started a voice portal for Parents and Students. They can listen Grades, Attendance status and other relevant information over *phone*. Please read features below and to listen Demo IVR call at *00 92 42 8315427. *Initially we are deploying it for a School and planning to spread it for Colleges and Universities. *Your comments and feedback are valuable to us.* *TeleKaam

Hello All One of our client Bank has 900 employees working in different locations. They need to record all internal and external calls. Can any body suggest Call Recording Solution for this requirement. We need to know the Hardware / Bandwidth and all requirements and costing. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006

Dear All, We are trying to install CDR Stats module. We are able to open its web pages but unable to retrieve data from CDRs. Can anybody suggest that how to connect this module with Asterisk CDRs database ? Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN:

Dear All, Can anybody please refer to Asterisk / Linux server maintenance checklist or tutorial ? Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 32801143 Email: kashif at haditelecom.com MSN: kashif__naeem at hotmail.com Gmail: meet.kashif at gmail.com Skype: kashif.naeem -------------- next part

Hello All We purchased *25* new AtCom AT- 530 phones. Four of them did not work for even once and some of them lost configuration after some days. I talked to AT Com people over chat for support. They have just 1-2 people for support who are also busy in some other activities due to which they are unable to communicate in proper way. Often support guy left conversation suddenly and is unavailable

Hello All We have developed an application for Home Delivery Restaurants using Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If someone is interested then we can provide him more details. - POP up window with caller data containing his/her name, address and transactions history. - In case of new customer, Pop up window with blank form to add customer data and

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Greetings Asterisk users! I'm trying to setup Asterisk system to act as a callback system together with callcentric (http://callcentric.com) but it appears that I hit common DTMF issue and I want to workaround this problem. Basically my current setup is the following: 1) I have dedicated Asterisk server that it is linked to my callcentric account 2) I have US phone number (DID) from

We have following problem. On some of the extentions I call cell phone after 10 seconds or so.Or, like this one below- we call cell and office phone at the same time ;Eric on extension 105 exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)         same => n,VoiceMail(105 at default,u) Where problem comes in - if person not at the desk - his cell phone shows call from OFFICE number and

Hello everyone, I want to connect a fax to an FXS port (TDM420P). For testing purposes, I connected an analogue phone to it first. However, when I pick it up, I cannot hear anything at all. The power cable is plugged into the card, the port is configured to use fxo-signalling. Also, immediate=no. Here's the files: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

I would like to download climate data files from PCMDI website using R. I tried this line below and I was not able to get the file mainly due to user name and password requirements. I am looking for help for setting up the user and password within R (or somewhere). I have read the FAQ but unfortunately I am a newbie on R and couldnt figure out how to do it. Many thanks in advance

Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for most / 90+% of

Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone.

what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/13145555555 at callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and

Hi Mohammad, A lot of time is being spent in addressing metadata calls as expected. Can you consider testing out with 3.11 with md-cache [1] and readdirp [2] improvements? Adding Poornima and Raghavendra who worked on these enhancements to help out further. Thanks, Vijay [1] https://gluster.readthedocs.io/en/latest/release-notes/3.9.0/ [2] https://github.com/gluster/glusterfs/issues/166 On