similar to: SIP server behind NAT

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk

READ READ READ .... http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Sun, Feb 22,

Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

Hi All; Really I need to know why when using the "h" in the exten =>, then we use DeaAGI with it? I am using vicidial and I see this line alot, so I need to know how it work (when it will be executed): exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) The question is: When

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Please cc me on replies. My employer recently upgraded to W2K3. I have no control over the employer's set up and limited access to information. Under the old server, everything was working fine. Now I can't mount the shared drive anymore. I'm running Debian sid; samba 3.0.6-3. ################################################ # mount shared_drive cli_negprot: SMB signing is

I am sure I'm doing something wrong here but not sure what. Our system administrator recently installed UnixODBC and the MyODBC driver on a Linux box running Linux version 2.6 x86_64. I have an .odbc.ini file in my home directory with following lines: [mydb] Description = MySQL server on my-server Driver=/usr/lib64/libmyodbc3.so SERVER=my-server I can successfully do the following:

I think I figured this out. I should not have put the Driver name in braces. Changing it from {MySQL} to MySQL seems to work. -----Original Message----- From: Marcus, Jeffrey Sent: Tuesday, January 12, 2010 6:09 PM To: 'r-help at r-project.org' Subject: Problems connecting with MySQL using odbcDriverConnect (RODBC package) on Linux I am sure I'm doing something wrong here but not

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP Could you please suggest transcoder to use from G711 and G729 and which is comptible with Asterisk. We will like to avoid using TDM if possible Also i remember that initially we didn't have G729 and were using only 711 for with vicidial but then also we had same problems. at that time it was only 2

Hello, I've installed VICIDIAL per the instructions on the astGUIclient website. It appears everything is working correctly. All the conference rooms have been set up, the database is running, and all the astGUIclient/VICIDIAL scripts are running. I'm using the VICIDIAL client on windows 2000, and it also appears to be working correctly. I can log in with no problems with the user

I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not even a call. Ring? Beep? Ping? Some sort of "hello world" connection. 192.168.1.1 netgear router 192.168.1.2 asterisk (vicidial) 192.168.1.3 ubuntu client 192.168.1.4 mac OSX client (not shown) Do I have a firewall problem which would impact a soft phone from establishing a connection?

All, I''ve upgraded to 2.0.2, and I can''t get my login screen (the first POST request in the application) to work. When I post this form, I see the "InvalidAuthenticityToken" error. I have protect_from_forgery :secret => ''my_secret'' set in application.rb and I am using an active_record session store based on this line in environment.rb:

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38