search for: tareksawah

Displaying 10 results from an estimated 10 matches for "tareksawah".

2010 Apr 10
2
Sending RTP media to a different server than SIP Signaling
Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _________________________________________________________________ The
2010 Sep 15
3
Skip Busy Agents/Channels from Queue
Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory
2011 May 02
3
out of the blue one way audio
Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet
2010 Sep 23
4
Asterisk and Digium TC400B
Greetings, Because of the heavy load and the high expectations of an asterisk server offered as a solution integrated with our CRM software.. we were looking into other possibilities than software Licenses for G729 and G723 codecs.. to lower the pressure on the processor giving it more space to do more work. We heard of a hardware (PCI CARDS) can be used with Asterisk that does the work. And we
2010 May 19
1
Asterisk and RFC 3261
Greetings List,Trying to interconnect with a new provider.. the require a?compliance?with?RFC 3261 ?so knowing less than needed about RFC documentations.. i would like to know if Asterisk is actually in compliance with?RFC 3261 or not..?Can any one help with this? Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
2009 Mar 12
4
Serving 120 concurrent calls
Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount
2010 Jun 28
2
restricting sip users to a certain useragent
Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will
2010 Feb 01
0
Asterisk for productive Calling Card System
Dear List, i have been thinking of building a calling cards solution based on Asterisk and a2billing.. i have a few questions regarding this solution and was hoping you may have the answers and could be generous enough to offer them. the servers i'm thinking of are with the following Specs: Processor: Intel X3210 Ram: 8Gb HDD: 2x500 GB Sata Internet Link: 100mbps Dedicated was thinking of
2010 Apr 29
1
Strange Invite issue
Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is going on? -- Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308
2013 Oct 11
1
GSM to SIP Adapter
Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one SIM card). any suggestions? Tarek Sawah -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131011/794f5a49/attachment.html>