similar to: Use ZAP/Dahdi channel for outbound only... no inbound?

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this

Greetings- First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I'm hoping they can be of assistance. I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID's coming in via PRI, SIP,

Greetings all- During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? Thanks! --Tim

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

? Hi, From asterisk 1.8, the CDR table is not logging the unanswered or extn busy calls which hit while in the queue. I am talking about this setting in the cdr.conf : ; In brief, this option controls the reporting of unanswered calls which only have an A ; party. Calls which get offered to an outgoing line, but are unanswered, are still ; logged, and that is the intended behaviour. (It also

Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. Are there any ways to tweak this? Thanks, Gavin.

Hello,I was using the ResetCDR() right after my IVR answered the calls to set them unanswered until or if somebody pick up the callI just upgraded to version 1.4.19.1 and now ... when the ResetCDR() function is called it seems to just erase the recordIs there another to set the status of the call unanswered ? is it a bug ?Thanks, -------------- next part -------------- An HTML attachment was

Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 "Wildcard AEX410" *53 FXO FXSKS

I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel <x> Or: Action:

> > Funny. The first thing I always do after a reboot is call in from my > cell to make sure things work. But last night I rebooted and immediately > tried dialing out (with a TDM842B) and got: > > WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to create channel of > type 'DAHDI' (cause 34 - Circuit/channel congestion) > > Restarting all of Asterisk or

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,

Because of a kernel upgrade i needed to recompile DAHDI. Dahdi 2.0.0 with 2.0.1 was working ok, after the reboot and compile it doesn't start anymore. Firstly it tells me it is using /etc/zaptel.conf which is deprecated, if i remove that file it complains that this file is missing..... Secondly i get an error about a missing dev file which refers to zap (which seems strange to me because

Dear friends, ? I am not able to capture the CDR records for unanswered calls. Only one record per call is coming in the CDR table. In the cdr.conf, I have enabled it by setting : unanswered = yes ?

Hi all These questions were asked back in 2009 at lists.digium.com and got unanswered: - Has someone been successful in using IMPORT on a Local channel ? - Is there a known limitation in doing so ? I run into the same problem. ${IMPORT(Local/1234 at example-abcd;2,CALLERID(dstchannel))} returns nothing. But I can read the dstchannel for it into the CDR. Asterisk is 11.7.0~dfsg-1ubuntu1

getting some errors on my test system. this is 1.4 (Asterisk SVN-branch-1.4-r214194) with a 4 port T412p card. Three of the ports are connected: Span 1 to the PSTN on a 10 channel bearer line, ports 2 and 3 are cross-overed (!) to each other. Port 4 is not plugged in. This has been working fine for several months. I updated a few days ago to the latest 1.4 branch. However, now I cannot dial into

Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run "pri intense debug" and make a call to the BRI span, I can see a

What is asterisk's behavior when faxdetect=outgoing in zapapata.conf? Does it turn off echo cancellation? Does it also change the priority to "fax" in the "outgoing context"? Thanks, Vieri

Hi, I've noticed that a small but meaningful quota of calls from my Alcatel PBX to Asterisk are failing. This does not always happen and it is not easily reproducible but on high traffic I do get a large number of cases. Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145 over a PRI E1 link. I see this in the Asterisk log: Dec 14 14:10:31 VERBOSE[11378] logger.c: --

I am using an 8 port tdm card and also I implemented a dialer using a .call file generator. As you know on the .call you specify the channel to call and then the contex/extension/priority to let dial plan continue when the call is bridge. My actual problem is that when the call process starts, asterisk (DAHDI) sets the channel as answered when the truth is that on the other side the channel has

So I hacked up the Dial app to return a numeric return code instead of changing contexts based on a number being busy or unanswered. The purpose for this modified dial app, which I call AGIDial, is to help me concoct a "follow-me" type of application. The app returns -1 for a completed call, 0 for unanswered, or 1 for busy. Well, I hooked the thing up to an AGI script that uses perl and