similar to: Problems with incoming PSTN calls

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

Hi all, I have a hangup problem when i get incoming calls on my ISDN interface. I use ISDN network controller [HFC-PCI] and asterisk with florz patch. Logs when the hangup happens follows: May 6 20:52:47 NOTICE[11532] cdr.c: CDR on channel 'Zap/1-1' not posted May 6 20:52:47 NOTICE[11532] cdr.c: CDR on channel 'Zap/1-1' lacks end May 6 20:52:47 DEBUG[11532] pbx.c: Expression

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Hello, I am having an issue whereby calls are being dropped randomly. I have an ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk install is based on Trixbox 2.0. However, I have updated the source code to the following. The Asterisk release is asterisk-1.2.20. Zaptel release is zaptel-1.2.18. And libpri release is libpri-1.2.4. I have include an extract from the Asterisk log

> > Hi, Remove usecallingpres=yes busydetect=yes from your zapata.conf file. and the restart asterisk. Hopefully you will not faced drop call issues. Regards, Vidura Senadeera. Message: 3 > Date: Mon, 24 Sep 2007 12:29:40 +0100 > From: "Richard Young" <Richard.Young at intrintech.com> > Subject: [asterisk-users] Asterisk Dropping Calls > To:

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

Hello, This is my first asterisk installation, and having read up on the documentation, and trying several tutorials i'm unable to get my outbound route working. I'm certain it's an issue with my configuration and my inexperience with asterisk. So i have my POTS phone connected to my digium card, and when i make a call, I receive the "cannot be completed as dialed" message.

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to

Dear all, When I make a call using my PRI line, all goes well, but suddently the call hangs up. I searched the asterisk logs, and I found that. Write to 55 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) What does this mean? Why this occurs? How could I solve that? Someone could tell me if it was a primary error (the primary shows red alert in all its channels) or it could be a

Hello, I'm using Asterisk@Home. I'm still new to Asterisk, and trying to grasp it all. I'm wanting to do a simple setup of One SIP provider (Broadvoice) and 3 SIP Software Phones. I'm able to call the VoIP provided line fine and get dropped to the digital receptionist (or mailbox). However, when I try to send outbound calls I get "Error 503 Service Unavailable" and

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on

Hi, I'm currently running CVS-HEAD 2005-09-03 I do plan to upgrade to the newest version, but need to do some testing with it first. In the mean time... does anyone know what these messages below are about? I've never seen it before, but when it happened it locked Asterisk up pretty good. Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on Feb 10 10:16:57

Hi, I've got an Asterisk system that I've just added an X100P card to. Incoming calls route to my call group just fine. When I make outgoing calls by prefixing with 9, they route to the PSTN network okay, get answered but then drop straight away. Has anyone seen this before and found a fix? I'm in the UK using a phone line from NTL. I've Googled for a bit, can find others

I am trying to configure Asterisk for BSNL, india network. I have successfully configured it for outgoing calls. When any outside number make any call to trunk then it receives the call properly but when the call is disconnected by inside extension then outside phone does not get a busy tone. Asterisk incoming call log: -- Executing [s at incoming:2] Dial("Zap/4-1", "Zap/1")

[also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling