similar to: background / goto commands

Hello- I am currently testing with a carrier that seems to be having some trouble around toll-free (800 number) access. While a problem, its the resulting behavior that I'm finding disconcerting. When I dial an 800#, I get the following response: -- Executing Macro("SIP/2700-e10b", "carrier-out|18005558355|70|r") in new stack -- Executing

thanks to those who replied. I have managed to get the functionality I was looking for working with a series of Macros. However, it doesn't work as simply as I would like. There are two issues I've run into: (1)Goto provides no way to pass variables between one context and another. (2)I can't find any way to Goto a specific point within a Macro when calling it. Mostly this is a

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.

I think the problem it's in your dialplan, extensions.conf: ; voicemail management [voicemail] include => misc exten => 6245,1,VoiceMailMain2() exten => 6245,2,Hangup() Check the last line, I have the same problem and was because I wrote wrong the Hangup instruction... Regards! Date: Thu, 10 Jun 2004 09:54:32 -0600 From: Chris Hirsch <chris@base2technology.com> To:

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

I am trying out a new service from www.talkn.com. They use Sipura to terminate your service like most providers. They are looking at directly connecting into asterisk in the future. Right now my configuration is Talkn?Sipura?Asterisk/FXO Card. When someone calls in and get?s answered and when either party hangs up , * releases the port with out a problem. If someone calls in and * answers the

i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. I have the following in extensions.conf: exten => 2100,1,Answer exten => 2100,2,MusicOnHold(default) and have uncommented the "default" line in musiconhold.conf:

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

I would like to access VoiceMailMain2 skipping extension and password prompting if calling from a resource that has a mailbox defined. What variables can I use to retrieve the calling channel & calling extension (if it exists)? Here is what I'm trying to accomplish (of course ${CallingResourse.MailBox} is not a real way to retrieve this info)... exten =>

in my setup i have several users with DID lines coming in from various sip/iax providers. within our old phone system, a user could call their own DID line, then hit the * key when they hear their voicemail greeting and be prompted for their password. is there any way this could be replicated within asterisk? i'm having trouble figuring it out since it steps through things sequentially,

Voicemail.conf in MySQL is not functioning where I get the following error from Asterisk messages log file: CLI debug output is as follows: Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack -- Playing 'vm-login' -- Playing 'vm-password' -- Incorrect password '1234' for user '0' (context = <any>) -- Playing

Hello- As part of the asterisk build/installation instructions it mentions that the zaptel drivers should be built and configured first. My question is whether they are required at all, in the case of a system with no hardware cards at all (as is the situation in my case). With them loaded I continually get the following message on my console (server not asterisk): Zapata Telephony Interface

I am having problems retrieving voicemail from outside the asterisk system. My extensions.conf is configured as follows: exten => 7900,1,VoiceMailMain2(s${CALLERIDNUM}) exten => 7900,2,hangup exten => 7902,1,VoiceMailMain2 exten => 7902,2,hangup exten => 7999,1,dial(sip/7999,20) exten => 7999,2,voicemail2(u7999@incoming-pri) exten => 7999,102,voicemail2(b7999@incoming-pri)

Hi guys, I'm running Asterisk-0.5.0 and accidentally stumbled on this problem while in the VoicemailMain2 application: If you login to it, or even if you call it w/ 's<extension>' to skip the login and press an '8' near the beginning (and possibly at any point, I'm not sure), the channel seems to lockup, even if the handset is hungup, the channel remains frozen

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

> Well that didn't work....I now get this error > > > Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to > create > channel of type 'SIP' > == Everyone is busy/congested at this time > -- Executing VoiceMail("IAX2/iaxfwd@65.39.205.121:4569/5", "b") in > new > stackJan 12 16:56:21 WARNING[4989]:

Hi, compiled 1.0 on macosx latest (10.3.5). compiled fine. when running, complains about voicemail2 module. Any hints? Marc. == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-09/23/04-09:20:48, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com>

Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody

To the list ... I got the IAX2 stuff simplified & working (for now). See my earlier posting to the list. Now, here's a question for you all. I found a posting by J Todd where he gives BW utilization for various IAX2 codecs with trunking on. Now, the number of calls I can sustain over an IAX channel, obviously is going to be determined by the capacity and state of the physical pipe.

Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum.