similar to: I guess my server capacity is ok

So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message: ZT_CHANCONFIG failed on channel

I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all went okay. using sangoma a104dx on both machine. I followed the write up on http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup I have the cross over cable between them. however, wanpipe shows connected but the signaling link does not align. i have my configs for host A ##wanpipe1.conf [devices] wanpipe1 =

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

do you have selinux enabled? It should not be. p p.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything. From: "Lachek Butalek" <lachek@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date:

Situation, simple home setup: * Trixbox 2.0 * Feature Codes installed * GNet PA-168V based ATA * Cheesy cordless analogue phone >From what I gather, dialing *70 from the handset should activate Call Waiting. All it seems to do is change the message "The person at <extension> is on the phone" to "<ring> <ring> The person at <extension> is

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the

Hi all, I have a "Cologne Chip Designs GmbH ISDN network controller" and I want to terminate voip calls via this ISDN card. My question is: How I must to wire the ISDN equipment with my ISDN card? With normal cable or crossover? How I can to check if ISDN card is linked with ISDN equipment? In this moment I have 1:1 cable between ISDN's, the mISDN is installed and "misdn

hi does anyone have an idea how it could be possible to do email -> fax gatewaying with asterisk + app_txfax, but still keep track of who sent the fax? i've thought a little about smtp auth, but it doesn't look too easy to integrate smoothly with asterisk.... roy

Hello list, I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -----ip network------- [Asterisk]----ip network -------[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. -- ------------------------------------------------------------

Hi List, I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/number@gateway) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes,

Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Did anyone try to install Asterisk on the Mini-Box M300 with a Versa mini-ITX board 1GHz VIA x86 CPU? The box looks promissing, but I am not sure if Digium cards are compatible with the mother board (Versa mini-ITX) Also I am not sure if the 1GHz VIA processor can handle a Digium 24 port analog board, or an E1 digital board. If anyone had tried the Mini-Box, the processor, of the mother

Hello everyone. I've got a HFC ISDN card that I'm using with chan_misdn and it basically behaves like crap. Echo is waaay worst then echo I get TDM400 card, sound is "choppy" (there other side is allays complaining about sound interruptions) and to top it all it detects fake DTMF's all the time. Is this a chan_misdn problem or is it a card problem? I really need to get