similar to: Slow dialling from PBX into * via E1

Hi! Could someone give me a hand? If I dial 200 for echo testing it works... Everytime I dial an extension ex. 505 get the error below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing

http://dovecot.org/releases/1.1/beta/dovecot-1.1.beta15.tar.gz http://dovecot.org/releases/1.1/beta/dovecot-1.1.beta15.tar.gz.sig It's been a while since the previous beta, so I guess it's time for a new release. I wanted this one to be called v1.1.rc1, but Squat indexes still aren't completely working. My simple tests seem to work OK, but after doing some random

Hi, Just a quickie - if I want to implement an * solution purely for voice (well, and physical fax machines / dialup modems..) on EuroISDN E1s, is there any benefit to the A104u over the A104c? I'm just trying to decide if the extra ?200 for the A104u is worth it :) Cheers, Gavin.

Hello Dovecot users, I finally managed to upgrade the ManageSieve implementation to dovecot-1.1. This also resulted in major restructuring of the code. The actual ManageSieve implementation is now available as a separate package and the patch now only contains the changes to the dovecot-1.1 tree that are necessary to use the ManageSieve service. The patch and the new package no longer

Hi, For some reason, I seem to have issues with dailing toll free numbers and can't seem to find out why, sometimes, I get a busy signal. Some other times I get weird errors from the phone. The error below was a simple busy signal. Here's couple of my info relevant to the problem: -- Reconfigured channel 1, PRI Signalling signalling -- Reconfigured channel 2, PRI Signalling

Hi I have a cisco 7960 Phone that connects to my Asterisk server without a problem. But when I call the echotest it just hangs up, echotests from other VoIP providers works just fine. I have tried a softphone and it works just fine. The error I get when the 7960 calls is this: -- Executing Playback("SIP/2000-180c", "demo-echotest") in new stack -- Playing

This 2 line code is doing what I wanted. exten => 200,1,voicemail(200) exten => 200,2,Hangup What I've been told is that they want the 20 year old phone system to light up the message bulb. (yea, a filament bulb, not an LED) To do this you pick up on the line that goes into Asterisk and do a: exten => 200,1,SendDTMF(200w#86) But I don't know the path to take to get that

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hello, I have the need to connect 2 remote site with vpn, the windows pc of the 2 site it can share the HD and printer. This is my configuration : LOCAL NETWORK A : ip from 192.168.10.2 to 192.168.10.99 | | | | eth0: 192.168.10.1 FIREWALL A : ( with debian ; openvpn ver. 2.0.beta15 ; shorewall ver 2.0.11 ) eth1 : xxx.xxx.xxx.xxx ( pubblic ip address ) | | | | INTERNET | | | eth1 :

Hi, I've been using 1.0 but moved to beta to see if it works before release. Both b14 and b15 dies sometimes when accessing folders. Last time I fixed this by moving one message from folder to another and then back. Next I'm trying to build this with debugging enabled. Tomppa > Feb 14 11:52:33 dovecot: [ID 107833 mail.crit] IMAP(tomppa): file message-parser.c: line 674: assertion

On Wednesday 03 August 2005 17:33, Jens von B?low wrote: > Gavin, > > >> Any ideas/advice would be warmly received right now! > > You are not going to like my response... Erk :) > The only way I could get this to work (luckily I had 2 identical sites and > was busy with the upgrade to the gen2 card) was to downgrade to zaptel > 1.0.7. Alas no - just moved down to

Hi, I've used Agents + Queues before with success, but I can't figure out why this trivial setup is not functioning... stage*CLI> show agents 1306 (gdh) available at '1306@internal' (musiconhold is 'default') 1 agents configured [1 online , 0 offline] and the internal context is simply: [internal] exten => _13XX,1,Dial(SIP/${EXTEN},,h) Now, taking this

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not

Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten => _811XXXX20,1,Goto(C-Internal,100,1) exten => _811XXXX21,1,Goto(C-Internal,200,1) [C-Phibee] exten => 100,1,Ringing exten => 100,2,Wait,1 exten => 100,3,Answer exten => 100,4,Dial(SIP/201&SIP/200,30) exten => 100,5,Hangup exten =>

Hi to everybody, I have the problem that nearly no information are displayed on the Asterisk CLI (asterisk -r). In former times (before patching Asterisk 1.0.2 with ast_data 1.0.2) it looks e.g. like this: --- snip --- -- Registered '96' (AUTHENTICATED) at 212.202.169.118:4569 -- Accepting AUTHENTICATED call from 212.202.169.118, requested format = 1024, actual format = 1024

Hi all I am in this strange situation: we had ser configured to relay calls to numbers to asterisk extensions and all used to work nicely, with both ser and asterisk running on the same machine with public ip (ser on port 5060 and * on 5061). We had to move temporarily our server to another provider which put our server on a dmz, so that now we have our server with private ip but reachable from