similar to: Asterisk with SC440 Dell(Big Problem)

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

For anyone looking for a cheap home ZFS server... Dell is having a sale on their PowerEdge SC440 for $199 (regular $598) through 11/12/2008. http://www.dell.com/content/products/productdetails.aspx/pedge_sc440?c=us&cs=04&l=en&s=bsd Its got Dual Core Intel? Pentium?E2180, 2.0GHz, 1MB Cache, 800MHz FSB and you can upgrade the memory (ECC too) to 2gb for 19$ bucks. @$199, I just

Does anyone on list have experience with Digium hardware in the following servers: Dell poweredge SC440 IBM xSeries x226 Have just had major hassles getting TE205P ISDN cards going in these boxes. No joy so far. Anyone managed to do it yet? Thanks. Mark

What cpu/memmory configurations have people had good luck with for a small office asterisk server... (polycom's/poe linksys) Dell's got their $200 SC440 going again until the 12th... and I'm thinking it might be just the ticket...

Hi all. I have a Dell SC440 running Centos 4.4. It has two 500GB disks in a RAID1 array using linux software raid (md1 is / and md0 is /boot). Recently the root file system was remounted read-only for some reason. The logs don't show anything unusual, presumably the file system was read-only before anythng was logged. Running dmesg showed this error repeated many times: EXT3-fs error (device

Hi! I wanna know if can run my AGI scripts as fastAGI scripts in Windoes server. I need a lot of script done in perl and I wanna move to windows server. I checked Asterisk::fastagi but I see that everything is for Linux. Somebody has idea to do this in perl. I dont want to change the language. TIA *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase

Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer

Hi! I have playing my icecast using Ice for Vorbis and ezstream for mp3 , of course , one by one. Now my question I want to have a few channels with different kind of music. How can I do this? Can I have a few mount point into ices or ezstream or can I run a few instance with different config files? I try this with ezstream but it doesnot work. Any cluees ? Edwin Quijada

Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [s at from-pstn:1] Answer("DAHDI/7-1", "") in new stack -- Executing [s at from-pstn:2] Playback("DAHDI/7-1", "vm-intro") in new stack -- <DAHDI/7-1> Playing 'vm-intro' (language

Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram.

Perhaps someone with more experience can help me solve this puzzle: Asterisk 1.4.18 on a Dell PowerEdge SC440, CentOS 5, 2.6.18-53.1.14.el5, Digium TDM400 with 3FXO(1-3) and 1FXS(4) Phone service is provided by XO Communications via their XOptionsFlex service (5 analog lines, 3 of which are used by Asterisk). For simplicity I refer to them as xx43, xx44, and xx45 - based on their phone numbers.

Hi all, I've just built a new installation of CentOS release 5.3 (Final) and have installed both <http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz>Asterisk 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing complex - Pentium Dual core 2ghz - 1gb ram - 70gb

Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT & T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look

I have a connection of Asterisk with Avaya by H.323 and so far everything worked well because only sent to Avaya. Now, the matter is that from Avaya will send me an IVR calls to capture credit card information, the link is active on Avaya 23 channels which is not how to configure Asterisk for those 23 simultaneous channels of Avaya's collect asterisk. Do not know if I can be with a group or

I don't know if this is a Samba question or an OS question. I'm working on an embedded project (microcontroller, limited RAM) that reads files from a FreeBSD/Samba (FreeNAS) box using TCP/IP transport on port 445. They're MP3 files so they're several MB at least. If I request 512 or 1024 bytes each time, I get 1024 bytes back from Samba every time. That's good. But, if I

HI all: Iam planning to use my asterisk box as callcenter?,any one can advice me with the best callcenter open source program based on asterisk . ? Any help will be apreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100307/116f1b75/attachment.htm