similar to: reliable capacity for a single * box

To the list ... I got the IAX2 stuff simplified & working (for now). See my earlier posting to the list. Now, here's a question for you all. I found a posting by J Todd where he gives BW utilization for various IAX2 codecs with trunking on. Now, the number of calls I can sustain over an IAX channel, obviously is going to be determined by the capacity and state of the physical pipe.

Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Thanks, Daniel

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

I have a nice used zhone channel bank I want to experiment with, but need a T1 interface for my * box to do so. The TE410P looks nice, but more money than I want to spend to experiment, and I don't need 4xT1, only 1xT1. Are there any good 1xT1 PCI cards that are recommended? Regards, -Dorn

Would anyone care to offer opinions as to the FXO interface which sucks the least :) I have an application in which it appears I must route certain calls out an analog PSTN line. Presently, I am testing an SPA-3000, but I can't seem to get the echo heard on the IP end of the call down to a non-annoying level. Any suggestions welcomed :) -Dorn

I'd like to get VM_CALLERID to include number in addition to name since often when calls come from cell lines or various other, the name is just a city, state and the number would be more usefull. Is there a way to get the number in the VM_CALLERID string, or is there a second variable I can use in formatting email vmail notifications to get the number? Regards, -Dorn

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Hi All, I'm very interested in real world experience of double digit number of users sustaining good quality audio in a single meetme conference. Personally, I have seen 23 users in one conf room, all coming in SIP, ULAW. Server is 3.2GHz proc, 1Gig RAM, 1-2 % proc utilization under 23 user load, perfect audio. I'm working on a conf bridge for 150+ users, could use some advice, if

In Timo's thread about RAM today, I noticed dmidecode and I got the data for my Dell Dimension 2400 (Celeron CPU) box, which is below. I won this box in a raffle, during February 2005. After I got the box, I got conflicting information, from Crucial.com and from Dell Latin America, regarding the maximum RAM configuration. To be conservative, I bought two 256 MB DIMMS from Crucial.com, early

can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run "top" to check the system resources usage

Hello, I am trying to come up with a good way to enforce a limit on the number of simultaneous calls that can occupy an IAX trunk at any given time. I have searched around and so far can't locate a config option that would directly label a IAX trunk with a specific number to obey (is there one?). Based on examples for the SetGroup and CheckGroup commands, I am thinking of using SetGroup

I'm trying to figure out how much call load I can put on a Dual Xeon 2.4 Ghz Asterisk server acting strictly as an IAX2 call director, as show in the diagram below. The idea is that I have N number of gateway asterisk servers connected to the PSTN using T1 Digium boards. Then, I have M number of servers where my agents and/or telephone extensions (whether they are IAX or SIP

hello list, i need to setup an asterisk system with 5 ISDN trunks. i found C4 cards but they are very expensive. i found that if i use 5 AVM Fritz! cards it would be very cheap. i want to use 2 boxes. 3 in boxA +2 in boxB =5 isdn. and i want, this two boxs to work as a single box so that one box can share ISDN hardware from other box. this system will be serving a call center. currenly we are

Hello, Could anybody tell me if there is a Linux distribution (or Kernel version) that works better with Asterisk. I am newbie and I don't know if there is a preferred Linux/kernel version for Asterisk. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040904/8798ff25/attachment.htm

I'm having a problem with IAX running through an intermediate asterisk box. Perhaps a small diagram will explain the situation better: *A ------- [cloud (public internet)] ------- *B --------[cloud (private network)]----------- *C Asterisk server's A, B, and C, are all connected together with IAX All asterisk servers are 1.6.0.6 Server A and B are geographically close, but connected over

Hello, We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created. The rest of the week,

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

I am having trouble getting trunking to work between a couple of my servers. All servers are running 1.0.9 stable version, and are working perfectly. All have a Zaptel card of some description, so timing is not a problem. Each server has a definition for each other server, using RSA auth, and qualify=yes (just so I can see ping times). trunk=yes is on globally, AND in each peer/friend

Dear Royce; Did ur problem resolved? Because now I am facing same problem. It look like that it happens with IAX trunk only, but does not happen with IAX endpoints that registering (as trunk does not register, it sends the call directly). My initial analysis that one of the following can help to let the trunks talk: if there is an IAX endpoints registering to the machines, then trunk become

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729