similar to: voicemail, busy does not work?

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer.

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server

I've been continuing to experiment with MySQL. I'm having absolutely no luck getting asterisk to read voicemail configuration data and mailbox configuration data from mysql tables instead of from voicemail.conf. The default Asterisk setup that reads from voicemail.conf and extensions.conf works fine. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox Enterprise Linux box.

I've got one that I haven't been able to solve. Hopefully someone else has had this issue. I'm using the paging script in free pbx, which appears to: Send a sipheader autoanswer, Create a conferece Add the phone to the conference But if the user hits the page extension, all the phones auto answer, and if they hang-up before the phones join the conference I end up with dozens of

Hi All Just got a new quadBRI card and connected one port to our Old PBX. When I make a call from a sip phone to a phone number the phone rings, I hook up, and the call on the sip phone allmost imidialely disconnects, after a fiew seconds the "real" phone disconnects too. Here is a trace: -- Executing SetCallerID("SIP/cramer1-b718", "45") in new stack --

I have the following in extensions.conf: [global] MP3ROOT=/var/lib/asterisk/mohmp3 [default] exten => 1111,1,Answer ; Answer the line exten => 1111,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => 1111,3,MP3Player(${MP3ROOT}/sample-hold.mp3) The command that runs is: 14030 pts/0 S 0:00 /usr/bin/mpg123 -q -s -b 1024 --mono -r 8000

Hi all, I am using asterisk1.2(vicidial). I am using like pbx . In this how can I confugure the internal conference calls. suppose I have A,B,C,D,E users these all peoples should be internal conferece . for them i was give 101,102,103,104,105 extensions. For this scenario what can I do exact configuration in dialplan and any to edit confugration files please help me . and how can they cut the

Hi all, I want to initiate a call from the asterisk to an extension, where I will forward the asterisk side to another extension later (to the conference extension). I can initiate a call uning originate call from an extension to the desired extension, but it would need someone from the originator extension to answer the phone. How can i register an extension to asterisk where it

Hi! I'm trying to transfer a incomming call from a Queue to another extension. I'm receiving a call from a queue with the AgentCallbackLogin. The queu is as following: Queue(sales|t) Which should allow transfers. So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

I have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To:

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn

Hi all: I am trying to use the ddply function to estimate the mean of 'Total','Fry','Smolt' and 'Fry.Eq' columns without success. I have the dput of my dataset below. I wonder if someone can give me a hand with this function. # dput(winter) winter <-structure(list(IDDate = structure(c(37L, 48L, 59L, 62L, 63L, 64L, 65L, 66L, 67L, 38L, 39L, 40L, 41L, 42L, 43L,

Hi Scott, That #1073741833 is a register mask. They are treated as aggregate registers (essentially sets of registers), so if it includes R9D and R11D, it will be treated as being aliased with both. These separate defs are there because they reach disjoint registers. -- Krzysztof Parzyszek kparzysz at quicinc.com<mailto:kparzysz at quicinc.com> AI tools development From: Scott

Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after

I've had 50+ people download the web components, and other than reports of compile issues, I have not heard if this collection has worked for anyone. I do plan to keep updating the * applications and the web pages, but I have almost meet all of our internal requirements and wonder if anyone else is finding it usefull. My focus has been and will likely stay on the user interface, since I have

Hi All, I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions. But when I installed AMP and created new extensions, I'm not able to call those extensions. I get the message that the extension is busy and it is forwarded to voicemail. What am I missing here? The workaround I found is by modifying the