similar to: hang up when pickup analog phone

I am having problems getting incoming calls from the PSTN to route to extensions, digital receptionist and even voicemail. When I call a DID number for one of the lines, it rings twice then says: "Goodbye" and hangs up. (logs to follow below configuration info). When I dial 7777 it goes to the digital receptionist without any problems. The system setup is simple; I have 8 PSTN

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:

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.

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

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.

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

Hi all, I'm new in asterisk and I got to set up a dundi config for my work. I have 2 PBX for the test, the two PBX are in the same local network PBX A : 192.168.199.23 PBX B : 192.168.199.21 my config files : (on PBX B , the config files on PBX A looks like it) /etc/asterisk/dundi.conf [general] bind=192.168.199.21 port=4520 cachetime=5 ttl=32 autokill=yes entityid=00:30:18:4C:33:53

Hello, I have an Asterisk 1.2.10, with a TDM with 2 FXO modules, and 2 GSM Nokia32. I configured the TDM with loopstart signalling. For a few days, all works great: Nov 9 09:28:54 VERBOSE[19103] logger.c: -- Called g1/6XXXXXXXX Nov 9 09:28:54 DEBUG[19103] chan_zap.c: Exception on 14, channel 15 Nov 9 09:28:54 DEBUG[19103] chan_zap.c: Got event Hook Transition Complete(12) on channel 15

[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

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

I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From

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 are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity recorded in our full log. Could you help us to explain what had happened. Thanks. === my friend, 801, from his room did a test by dialing echo test in freepbx, *43: [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [*43 at from-internal:1] Answer("SIP/801-000003f5", "") in new stack [Aug 20

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

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

Hi, I am trying to dial # or *411, in order to understand what the * box should answer me. In both cases, I only ear "Good-Bye" (italian , "arrivederci") dialing # -- Executing Wait("SIP/555-a2e5", "1") in new stack -- Executing AGI("SIP/555-a2e5", "directory||ext-local|lo") in new stack -- Launched AGI quitScript