search for: tzompa

I'd like to have Asterisk log useful messages during operation. Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway. Thanks, Doug.

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne

Hello! I'm tryuing to bring up an R2 connection but eventhough I've followed the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems to be missing. When an incomming call is generated I get: Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 <- 0001 [1/ 1/Idle /Idle ] Nov 24 06:01:17 WARNING[-197416016]:

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

how to adjust voice volume for sipura 2000 and cisco ata186?

Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is when dialing to the Harris PBX, it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play a prompt as soon as it is picked up, the promt is beign played as

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

...300> host=dynamic nat=yes dtmfmode=INFO mailbox=300 disallow=all allow=alaw allow=ulaw allow=g723.1 allow= g729 Many thanks, Aisling. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alyed Tzompa Sent: 04 January 2006 00:28 To: asterisk-users@lists.digium.com Subject: re: [Asterisk-Users] confusion about contexts I'm a bit confused on how you get your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip user? in this case, which one?, if not is it iax or t...

Isn't your problem more about NAT traversal rather than the phones themselves? if so better use some iax softphone, have a look at: http://www.voip-info.org/wiki-VOIP+Phones of course you can use SIP based hard/soft phones but using iax based ones is cheaper and faster. Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Mon Oct

Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is that when dialing to the Harris PBX it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play it a prompt as soon as it is picked up, the promt is beign