similar to: Serving 120 concurrent calls

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

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

Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/816aecdd/attachment.htm

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

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

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>

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

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

This is the problem: Call coming into a queue in ringall strategy, if a member (SIP) of the queue is busy when entering the queue, and this member comes free after a little time, the member never rings.. How to solve this? I changed all parameters of the queue with no results... Wath i need: If one member of the queue is busy when a new call come in to the queue, this member can hangup and

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

Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/XXXX, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using

Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango

Hi I have On func_odbc [EXEC] readhandle=ressqlserver writehandle=ressqlserver readsql=${ARG1} writesql=${ARG1} I'm trying an update on dialplan: exten=> 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})}) On Cli: WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla set campo = 4356] Any idea why is this?? The query

I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090427/cc9690ee/attachment.htm

I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5 on DELL PE2950 and using ISDN-10. I thought about cutting over to production tonight when I noticed a serious problem. SIP calls are fine but if I dialed to outside (Dial(Zap/g1)) a few times or someone called in a few times, Asterisk just froze (cannot enter anything on the CLI console) and then even the machine had to be

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

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

I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

Hi All, I am newbie in this asterisk and a2billing technology . i had successfully installed asterisk in my server fedora -8 [server behind NAT/STUN] i after installation i can able to create users and tested the call features with X-Lite . the was working fine . after i installed the A2Billing in my same server with follow the steps from a2billing installation guide. but u cant access the