similar to: No correct IP in RTP media stream

I am so happy about learning how to read in multiple Excel files, that I have to try and make another improvement. I know what I have been doing is clumsy, but it works. Hopefully, someone can suggest a more elegant solution. As a novice, I have been using MS-Word and mail merge to write my code. I start with about 2 pages of code, and end up with 2,220 merged pages that I copy and paste into R.

Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =>usuario1:pass1@<public_ip_2> In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=<public_ip_1> dtmfmode=inband Logs in * are the followings In * one logs: Sip

Hi all, Now I'm working with a E400P, and I don't now if it's possible to do the following. I want that and AGI script (Perl) recieve a call, and the user introduce the date, the time and the destination phone number (where the temporized AGI must call). Before an AGI script will call to that number in the date and time introduced by user. That's possible, and it's how can I

Hi, On Tue, 18 Apr 2006, Autoegocrat wrote: > Ok here's the deal. I set up Icecast and Oddcast, got them to communicate > properly, but cannot access the live audio remotely. On other computers in > the LAN it works perfetly. So, I guess there's nothing wrong with your icecast or firewall config. > Currently I operate wirelessly behind a D-Link > 514 router (a mite

Hi everybody, Firstly I'm going to describe the scenario where I'm working. I use a E400P with Asterisk CVS-05/22/03-11:14:50, and I'm working with asterisk trow AGI scripts (Perl). The configuration of extension.conf is: exten =>_s,1,Answer exten =>_s,2,AGI,script.agi Inside the AGI script is call Dial application as follows: print "EXEC Dial

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24 network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is

Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the "flash" button instead of "transfer button" or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with "flash", the comunication between the person who is calling and is already in the queue and the target

I am trying to move from MS Windows (of which I am a std user) to Linux and, 10 days ago, I installed Mandrake 9.0, everything is OK except for WEB navigation: even if surely connected, I cannot navigate and the browser answer is always:...host unknown.. Apparently the problem is the firewall (Shorewall) installed by default that rejects all comunication, infact looking in the Firewall settings

I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091027/bb5e7b1a/attachment.htm

Hi everybody, Now I'm using SJphone on a win2k client an * as proxy SIP and GW to PSTN. I have doing some test, but I have the following question. It's possibles to make calls to external PSTN numbers without define an extension to make the call???? I will try to explain-me better. I have done some calls like sip:xisco@A.B.C.D, where in extensions.conf there are an extension like this:

I have sent a Polycom IP500 phone to an overseas remote user who has a Speedtouch adsl modem/router. The phone is connected to the 4 port router which performs NAT for the network. The phone rboots, find the server, downloads it's config and tried to register. Then it immediately becomes UNREACHABLE. Jul 26 17:44:41 NOTICE[21026]: chan_sip.c:9138 handle_request_register: Registration from

Hi. I install from source a wine 20090914. I have problem with printers. I have a cups printers in the system. When i open a Word it see a printers in the print box I can choise right one and I can see how it send task to the printer, but this is never print. I check the localhost:631 jobs and I don't see any task from wine, tahat I just send to the printer. In howto I found that cups

Hi everyone, I'm going to install an IAXy in my client's office but there the internet conection has a private IP address, however the Asterisk server has a public IP. Private IP Public IP (IAXy)--------(ISP LAN)-----------((Internet))-----(Asterisk Server) This setup can cause any problems to the comunication process? I'm aware that

Hey Folks....I am on the 1.2 branch with the latest from Subversion. I've been having a rough go for the last several months integrating asterisk with out Altigen system. I can get calls inward just fine. I have zero missed interrupts on the digium 110p card. I have zero frame slips according to both sides. Outgoing calls sometimes work, but more often than not I get the following: --

Hi! I want to comunicate with FLASh interface and a generic application in user space through swfdec. I see that there are some function of action script in swfdec that permit to load some variable in FLASH but....How I use it? Someboy could send me some examples pls?? And...for us, what is the best mode to create the comunication between "FLASH" - "swfdec-0.5.2" -

Hi I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1 The process of installation was the following: First I compiled and installed Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the ztdummy (modprobe ztdummy) and then i installed Asterisk: make make install make configuration make samples I started Asterisk, and created one SIP account, with the following

Hi! I have problems with my cellphone. It does not seem to quote the strings to the login command. I can logon using ie SquirrelMail but if i try to login using AndroidMail it just says "autentication failed". If i try maualy i get similar results. Changing password for test to "password-" a login test password- a OK Logged in. Changing password for test to "password

Hi all I really need some help here! All howtos seams to handle thing if one is going to install everything on one single computer. I've a fully functioning empty ldav3 Slackware server with openldap-2.1.25, Berkeley DB-4.2 ,mit krb5, cyrus-sasl, openssl. Everything seams to work fine. Now I would like to add Samba-3 on separate Slackware computer to the ldapv3 server. I've installed

Hi people. I'm trying to config a: MGE ELISE PREMIUM 1200 NT 02 Serial Nr. 92HE04008 With its serial cable into /dev/ttyS0. I had configured ups.conf, with no luck. Here is my ups.conf: [upscura] driver = mge-shut port = /dev/ttyS0 desc = "UPS" The permissions into /dev/ttyS?: crw------- 1 nut nut 4, 64 mar 8 18:43 /dev/ttyS0 crw------- 1 nut nut 4, 65 mar 8 18:42