similar to: call outside from FXS through FXO

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

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?

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

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside

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

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

> -----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'

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'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

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

Apologies if this is a repeat: I trawled through the archives and couldn't find a reasonable answer, so I'm asking here. I have an Asterisk install connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case, Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT device that connects to the Asterisk install, and using this setup I've been pretty

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

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)

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

I am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------

All, Alright, I've looked around the internet, the voip-info.org wiki, and browsed the contents of this mailing list. While I've found a couple of scenarios that are close to this one, I haven't found one that uses my particular card (T100P). Without further delay -- I have successfully configured internal SIP services between a Snom 200 and a Windows X-Lite client and have

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on