similar to: Asterisk incoming caller id problem

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

Hi to all I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus. I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start: [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When

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

Skipped content of type multipart/alternative-------------- next part -------------- asterisk1*CLI> soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --

I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for a queue up to 15 agents through a PRI line, it was working fine for more than 1 year, suddenly, when there is a load on the queue, the asterisk service disconnects and the calls are dropped. And the service starts again after few seconds, and so on. I am not using fax. I checked PRI by zttool and there are no alarms. The cdr logs

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings

I have installed my first Asterisk implementation using the Asterisk@home ISO. I am using the SJPhone software. Using the setup page, I have been able to configure two extensions. Whne I dial from one to the other, the other does not answer even though it is registered. Watching the log in the CLI, I can see that recorded messages are being played;: == No one is available to answer at this time

Hi I've been looking for 10 minutes and cant find dialparties.agi Can anyone tell me what folder this is located in as I'm going crazy. (if it makes a difference I use asterisk@home and am replacing the AMP dialparties.agi file) Super big TIA, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hellos, I have asterisk 1.2 and freepbx 2.3. I have edited the agi script(dialparties.agi). Everytime I restart asterisk, the file gets overwritten. How do I make sure my changes are not overwritten? What generates dialparties.agi? Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting 'transfer' then '801' then 'send' on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested

chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks