similar to: about call out : a strange question.

Specs: Fedora Core 3. Dual P3 600 (Dell PEdge 1300) SCSI Disks 1x X100P (channel 1) 1x TDM20 (channels 2+3) 1x Knockoff X100P (channel 4) I am looking to have all local and all toll free calls go outbound through the Copper line, and all long-distance and international to go out through the Vonage line. This way I can eliminate LD on my home line, and pay minimal LD charges through

I install ser and found my ser don't support mysql. my ser version : ser-0.8.14_src.tar.gz and ser-0.8.14_linux_i386.tar.gz who can help me? thanks. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

I use xlite , but it isn't support video when it is free. who used better softphone ? Thank u. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

I have a system include asterisk + ser. when I want to limit a dial out to pstn , I will do that : extensions.conf exten => _9NXXNXXXXX/myaccount@sip.com,Congestion exten => _9NXXNXXXXX, 1,Dial(ZAP/g2/{EXTEN:1},30,t) exten => _9NXXNXXXXX, 2,Hungup but I don't confirm is it right. I have no env to test it. who can help me?

ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

Two problems, possibly related: Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4 are FXS, 5-8 FXOs. Here are the config files: /etc/zaptel.conf: -------- fxoks=1 fxoks=2 fxoks=3 fxoks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 loadzone = us defaultzone = us -------- /etc/asterisk/zapata.conf: -------- [channels] language=en usecallerid=yes hidecallerid=no

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Matteo Brancaleoni > Sent: Monday, July 26, 2004 5:22 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based > PCIISDN card): Unable to create channel of type 'Zap'

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

I read the wiki and looked at the examples, but I'm still having problems. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

I'm not having any luck getting the ExtensionState action of the Manager API to work. The response is always success but the Status is always -1 which to me means an error. Here is a typical telnet session. ----------------------------------- stockholm:~ # telnet localhost 5038 Trying ::1... telnet: connect to address ::1: Connection refused Trying 127.0.0.1... Connected to localhost. Escape

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

I'm having a problem getting 3 way calling to work correctly using two outside lines and one extension. The two outside lines are connected to the X100P's and a standard model 2500 phone is connected to the TDM10. When I dial the first outside destination 9xxxxxxx, the call completes correctly. When I flash the hook switch and dial the second location 9yyyyyyy. The call doesn't

Hello All, I'm trying to dial out with no luck. I'm using Asterisk@Home defaults. I have one X100P card and SJPhone. *CLI> dial 96985628 No such extension '96985628' in context 'default' Here is my exten [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>