similar to: Messaging System

Hi to all, I need some help, I have an Asterisk Server in a small call center, for inbound calls I setup a Queue in queues.conf and their respective Agents in agents.conf, but when an Agent is calling out and a call is coming from PSTN the call is send to that agents which have a call in progress. How I can fix this in order to have only one call at a time. I think in limitonpeer and call-limit

Hi to all, I actually have an asterisk server configured to write CDR mysql records in the same machine (localhost), but I want to write this records to another machine also in mysql at the same time, It is possible? It means that I want save the records in both machines. Thanks in advance. Ricardo Melendez -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to All, I dont know much about PCI express slots in newer Servers, my doubt is if the Digium and Sangoma PCI express cards, are compatible with the x8 PCI express slots that come in the HP Proliant ML150 G5 server. Thanks in advance. Ricardo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

HI to All, I have an issue, when I connect a TDM04B and a TE212P on my server (Intel Entry Server board S3000AH) and boot RHEL4 the system launch this message: PCI: Unable to handle 64-bit address space for I think the problem is the TE212P because the TDM04B is detected as a Tiger Jet Network Modem. I have installed both cards in a PCI-X 64 bits Ing. Ricardo Mel?ndez Rosales Analista

I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by root@............... on a i686 running Linux. I just spent some time in testing this. I tested the local and IAX2 trunks. Both worked flawlessly. Any comments? Darren Wiebe darren@aleph-com.net

Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with "no, this is absolutely not for doing call marketing". I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I'm using the Local channel in an app of mine and I'm finding that the app is being cut out of the call path. You used to be able to avoid this using the \n command but that doesn't seem to work any more. This is on a recent version of Asterisk. Any comments/suggestion? Darren Wiebe darren@aleph-com.net -----BEGIN PGP

Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ

Hi, What do I have to do to get local\number to work in a context? It works from my [from-internal]... however from subcontexts it does not work: Jan 6 15:55:32 VERBOSE[20237] logger.c: -- AGI Script Executing Application: (Dial) Options: (Local/570323xxxx) Jan 6 15:55:32 NOTICE[20237] chan_local.c: No such extension/context 570323xxxx@default creating local channel Jan 6 15:55:32

Hi list. I have a fine running Ser+Asterisk environment and have just installed ASTCC. It?s working fine either, including its caller-id authentication feature (the one we pass the card-number as CALLERID variable and number-to-dial as EXTEN variable). The issue, a great one, is that when the credit is about one minute to end, the ASTCC prompt gets into the call, says that "you have one

Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1.

Does anyone have a mysql query that will compare a number from the asterisk cdr to a table of international country+city codes to determine the closest match? The two fields are; 1. Asterisk mysql cdr 'dst' field - sample record value '011441316551212' 2. rate table data like this DialPattern 011447977 011447979 011447980 011447981 011447984 011447985 011447986

Hi, I have a client who has asked me to look into the delivery of 30 second audio messages to a list of opt-in customers. Probably looking at about 5,000 messages a week over a 6 week period. I know that this would be a piece of cake to have someone develop but I thought I would ask here first if someone is already doing this and what they would charge to take this on as a hosted solution

Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090408/f6d5af5a/attachment.htm

Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on

Does anyone knows a working FOSS rating application for Asterisk? I tried CDRTool, but since it's free for non commercial use, it won't suit me. Also I gave a shot at Trabas (the one everybody says it sucks), but I couldn't understand how does it pull records from Asterisk's CDR table. The rate-engine addon (http://www.voip-info.org/tiki-index.php?page=Asterisk+addon+rate-engine)