similar to: free sip termination

Hello list, I have about 20 Digium TE-405Ps out in the field, and I started having trouble with one just recently. The card had worked fine for a month with 4 PRIs in NFAS configuration, and then all of a sudden I started getting a disappearing D channel. A restart of asterisk / ztcfg /module unload-load did not fix the problem, but a reboot [not power off, just a restart] would bring it

Here is some code to do authentication by caller ID for DISA through AGI. My original code had a bug in the Mysql query code, and there was a hangup in the wrong place [that's what I get for coding something at 2:00am], but the attached code works correctly. Take note of the REGEXP for the CallerID variable. When I tested the code from the PSTN it worked because there was no name component,

I'm having a problem with one of our newly built call processors. I'm using the stable branch of the asterisk code with a single TE405P on a P4 3.2ghz running kernel 2.6.6 with Hyperthreading and APIC enabled. The machine has 1GB of RAM and HIMEM is enabled in the kernel. Calls terminate properly and everything seems to work okay, but the system died after only being up one day with the

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get

GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can

I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the

Last tuesday I moved the asterisk server from 1.0.7 to 1.0.9, while leaving the zaptel drivers at 1.0.7 because it was a "lunchtime" update. This is a box with two TE405Ps in it, and all eight ports in use. Today I unloaded the 1.0.7 drivers and replaced them with 1.0.9 and oh boy... two of the 8 PRIs didn't want to come back, I got a million of FCS errors over the console and I got

Hello, I have two TE-405Ps that I am having trouble with. I'm using an Intel 865 motherboard with a Celeron D processor. Kernel 2.4.26, Slackware 10.0. my /proc/interrupts: CPU0 0: 172317 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 2003 XT-PIC eth0 8: 1 XT-PIC rtc

I have a little menu set up where hitting 1, 2, or 3 places the call through to a cellular phone over IAX. That works. However, if caller hits 4 to go into voicemail, the system hangs up. Am I doing something wrong in the dial plan, or is this a CVS change? I had no trouble with this until I upgraded to about 07/21 CVS, and I'm on 07/23 [latest] now with same results. My dial plan:

Hello, I have been running some tests to get a better understanding of PRIs and the HANGUPCAUSE variable and I'm not having any luck. I have tried calling disconnected numbers and the call is connected through to my extension and I hear the tri-tones. And it looks like HANGUPCAUSE is always 16 (AST_CAUSE_NORMAL_CLEARING). Am I doing something wrong, or am I just misunderstanding? Also,

I wonder if anyone can suggest ways to diagnose an infuriating problem being experienced by customers of a company I did a large Asterisk project for. First some background: The system is a conferencing system using a modified MeetMe. There are seven Asterisk boxes (we call them bridges) each with four T1 PRIs into a TE405P. No VoIP is involved. A conference is always local to a single bridge.

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Currently the site hands out a virtual 87820-xxxxxxx number but I intend to add the ability to get a free United States DID

Message: 5 Date: Thu, 19 May 2005 21:44:11 -0500 From: "Matthew Simpson" <matthew@txlink.net> Subject: [Asterisk-Users] cisco 7960 question To: <asterisk-users@lists.digium.com> I have a stupid question. How do you remove line presentations on a cisco 7960 ? I have 3 line presentations I don't use anymore and I can't figure out how to remove them. If you look in

BJ, >BJ Weschke <bweschke@gmail.com> >Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom >SIP termination vs. DS3 >To: Asterisk Users Mailing List - Non-Commercial >Discussion <asterisk-users@lists.digium.com> >Message-ID: <79cf63305051908056c284cc9@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 >Did I miss pricing/availability

The freenum.org beta continues to roll forward. If you have an Asterisk or SER SIP gateway/proxy, please see if you can make some sense of the examples below and install them in your system. Your users will hopefully be able to dial toll free numbers in various nations just like they dial regular numbers in those same country codes. I'd like to get some additional people trying to make

I have an Asterisk box with free local termination to area codes (305) and (786) [Miami area, US]. I want to configure it to accept incomming VoIP traffic (can't use IAX) and terminate calls over the PSTN network. I need help with the configuration and also some incoming traffic for testing purposes. Please contact me if you can help. Regards, Alejandro. -------------- next part

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hi List, I have configured TE121PF card in E1 mode. I am using asterisk 1.6 and freepbx 2.7. My card staus turn in to green and looks like sync with the service provider. My service provider is BSNL - India. I have one toll free number for incoming and one land line number for out going calls. Problem : If i am calling to the toll free number, i am getting the ring but that call is

Forget the sound card. It isn't related. The subject above should have read 'TE405P No Voice Problem' or something similar. It appears to be a zaptel timing issue, but I have found a workaround. For those of you just tuning in, here is the story: I have a CentOS/Intel 865 box currently running Asterisk 1.2.7.1 and zap 1.2.5, both compiled from the source available off the main