similar to: extensions.conf / test DID

I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command "sip show registry" and do not see it set up. I run sip show peers and I do see an entry. I have not configured anything other then entries in the

This seems basic but something is missing..... I dial from my cell phone to my DID and enter the context in extensions.conf I am hoping to cascade through the plan and successfully automatically dial the 1444 number listed. But it fails. And, I dpon't know why? Should I removed the Hangup application? Syntax issue somewhere? I have a good SIP registration with the vendor, voipvoip.

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Basic Dial Plan Why is this plan not engaging the line exten => 105,n,Dial(SIP/voipvoip.com/17035013333) and dialing the 703 number? The logs and debug dont show any problems.... [incoming] exten => 4444444444,1,Answer() exten => 4444444444,n,Wait(1) exten => 4444444444,n,Playback(beep) exten => 4444444444,n,Goto(105,105,1) ; ; [105] exten => 105,1,Wait(2) exten =>

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hello, I want that, when call is answered , callee and calling would hear different prompts and after promts the calls would be bridged. I've tried this situation: exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world) exten => s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno)) But these prompts play not in the same time: just after conf-enteringno prompt

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says == Using SIP RTP CoS mark 5 -- Executing [4567 at sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack -- Called trunk10 at 147.120.203.98/4567 [Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by

Hey, You know, the Digium guys said both are good. They said the the DNS method is better because you dont have the extra point of failure (SER) but said the SER method is better in that it gives you more exact control in the handling of the calls and registration. They did acknowledge there would be a possible downtime only for incoming calls to users with dynamic IPs if the

Dear all i'm creating an outgoing call to number xxx with this command: http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external &Exten=testDTMF&Context=cRETEUNICA&Priority=1 wich points correctly to this portion of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten => testDTMF,n,SayDigits(${digito}) The

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

Hi all, Seriously, I've tried to read everything I could find (& search for) on voip-info.org and other sites about this problem, but have been unsuccesful. Equipment: xten lite X100P Whitebox linux running Asterisk / AMP D-Link DI-804HV (VPN router) I have installed another DI-804HV at a second location and created a tunnel. For the computers behind that unit, everything works fine

Grrr. I'm using outlook web access and there's no way to do inline replies. Anyway... Gabriel. Using SER does not create a single point of failure. You install three SER boxes. Single point of failure gone. It does not take several seconds. If your phones are configured for SRV, and 2/3 of your SER boxes down, it takes about 2s. That's not bad for a 2/3 system failure. You can

hey all, after following the directions in the "FreeBSD Active Directory Domain Member Mini-HOWTO" http://web.irtnog.org/howtos/freebsd/winbind i am able to get my machine to the point where i can query users with 'wbinfo': $ wbinfo -u|grep galbrecht galbrecht i am unable, however, to login to my machine using any service, telnet for example: $ telnet -K localhost

> -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] > Sent: Thursday, March 16, 2006 8:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on > *HAandPolycomphone!! > > > > > > "Q: What are the plans for HA? > > That's BS. Last time I

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Hello Everyone, I would like to know if I can call one of the columns of a list, to use it as a variable in a function. Thanks in advance for any advice! Jana -- Jana Makedonska, B.Sc. Biology, Universite Paul Sabatier Toulouse III M.Sc. Paleontology, Paleobiology and Phylogeny, Universite de Montpellier II Ph.D. candidate in Physical Anthropology and Part-time lecturer Department of

I need predict to ignore rows that contain levels not in the model. Consider a data frame, "const", that has columns for the number of days required to construct a site and the city and state the site was constructed in. g<-lm(days~city,data=const) Some of the sites in const have not yet been completed, and therefore they have days==NA. I want to predict how many days these sites