similar to: quick question on conf bridge

We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Any thoughts. Jerry

Created this function on one of my machines today, thought others might find it useful: freesip() { comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr -d \]) | grep ^[[:digit:]] } On RedHat/CentOS based systems you can create the following file to have the function available on login: /etc/profile.d/freesip.sh # Free SIP extensions freesip() { comm -2 <(seq $2

Hello, I am in the peculiar situation to have to set up a PBX for two independent sites, but operated by the same entity. Yes, I could set up two VPSs and install Asterisk to each, put common stuff (e.g. conferencing setup) into Git and share between both using includes, but for various reasons (among them simplicity and cost), I'd prefer a single Asterisk instance. I know I can #include

Hi, I am running asterisk 1.6.1.6 with a howler screamer card. I have g729 and alaw trunks from a pbx /sip providers. The howler screamer will only transcode from g729 to alaw / ulaw but my voice prompts are in SLIN and throws errors when i try and access these applications. Is it simply a case of converting the prompts into other codecs and asterisk will pick these up? ? Thanks

How many simultaneous conference meetme setups can be supported in the same time on Asterisk, and what are the corresponding server's specs for this. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100414/ba26a927/attachment.htm

Hello everyone. I have server with 2E1 PCI card, asterisk 1.4.35, dahdi 2.4.0, libpri 1.4.12-beta2. One PRI trunk looks to PSTN and take a clocksource from telco. Another trunk looks to PBX with DECT system. Some outgoing calls from asterisk to PSTN drops. The last message that exists before hanging up process is: DEBUG[28467] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/... This

I have a TE210P digium card that has 2 E1/T1 ports. the code in my extensions.conf file for span 1 is : [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 ; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This

Asterisk segfaulted on me the other day; how do I tell it to generate a core file so -- if it happens again -- I can attempt to debug? I looked in the obvious places in "make menuconfig" and didn't see anything appropriate. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for PSTN-IP gateway. What is the maximum call handling capacity I can achieve with this server? I want at least 480 concurrent PSTN-IP calls. That mean I will have to install minimum 4 x 4E1 cards and run 480 G.711 RTP sessions. No call recording. No IVR. Pure gateway functionality. Can I achieve this capacity with given

with the slip of a finger, i destroyed by extensions.conf (grep -i > extensions.conf) I have a backup that is dozens of hours of code old. is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ?

Hi all, I'm going to configure a middle asterisk installation. I'll use a TE210P to connect a T1 channel bank and a PRI E1 line. I'm thinking on using a SuperMicro P8SCT Mother Board that has a 1x 64-bit 133MHz PCI-X 3.3V. In TE210P documentation I've read: The TE210P is a 32-bit 33MHz card keyed for 3.3 volt operation. Can SuperMicro slot (that is a 133Mhz slot) be used

Hi Everyone, Can you tell me how many concurrent TDM (Dahdi) calls that a single asterisk box can handle. Configuration is as follow : Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9 Also do you know a good tool to stress out asterisk? Kind regards -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* --------------

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

I have upgraded Asterisk from 1.4.22 to 1.4.30 and I have noticed I am getting a lot of errors like this on the console :- ERROR[23912]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe I have tracked it down to a perl AGI script which performs our own CDR recording. It is called before the start of the call, once answered and again when the call is hungup. It works fine when

I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go

Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin