similar to: incoming calls fall into echo test mode

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists & irc & Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

Hi I have a cisco 7960 Phone that connects to my Asterisk server without a problem. But when I call the echotest it just hangs up, echotests from other VoIP providers works just fine. I have tried a softphone and it works just fine. The error I get when the 7960 calls is this: -- Executing Playback("SIP/2000-180c", "demo-echotest") in new stack -- Playing

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

Any ideas ????? Jonn -------- Original Message -------- Subject: [asterisk-users] Internal LAN echo problem Date: Wed, 24 Oct 2007 08:34:32 -0500 From: Jonn R Taylor <jonnt at taylortelephone.com> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users

Hello, I installed 1.6.2.17 version of asterisk. Set the user database to realtime. I have no problems with sip users. They can register talk etc.. With iax clients, they can register also.. And when they call iax to sip, it works. When they make an echo test..no voice received on iax clients. When they make call from sip to iax ..no sound received on iax clients. I didnt see any clue on debug.

Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was "taken advantage of" over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

Hi, I have changed some prompts in /var/lib/asterisk/sounds and Asterisk still play the old ones, even if they does not exist in the file system anymore. There is any cache used to play prompts files? If yes, there is any way to purge that cache? Tried with 'reload' or to restart Asterisk without any luck. I have not tried and I don't want to restart the computer too. Thanks, Dan

I've installed successfully freePBX with Asterisk, and got various internal extensions working, however. recently my internet facing IP address has been removed by my ISP (for various reason) and I'm not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static

I have setup a asterisk with freepbx, a TE122 and i have an ISDN. My problem now is that callers are experiencing echo. checked on dmesg i saw this: # dmesg -c dahdi: Disabled echo canceller NLP because of CED rx detected on channel 2 i searched google but found no soolution and i have no idea what that error means. below are other details that might be of value. # dahdi_hardware

Hi All, is it possible to choose outbound route by checking the extension of the caller? e.g extension that starts with 3 goes to outbound route 1 extension that starts with 4 goes to outbound route 2. Basically, i'm hosting two(2) office, extension 3XXX is office 1 and extensions 4XX is office 2, they both have the same dialling pattern so i need to choose route based on source.

Hi, There is any experience using Asterisk with VMWare? I think about installing a virtual linux box over VMWare and then Asterisk over it. Thanks, Dan

The setup is a samba server with mixed clients (samba clients and windows clients). The problem, I want the linux client to mount there home to their home share on the server. The problem is, I have followed the guide mentioned below and everything works except that the linux usernames have the format domeinnaam+username as a result of which pam_mount wants to mount

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Hi all, In response to Rodrigo Rosas''s message about mb_chars.upcase not giving the expected result on 1.9, I''ve done some work in a fork to make String#mb_chars always return an instance of a proxy class, both with Ruby 1.8 and Ruby 1.9. The end result of the patch is (hopefully) to make Rails'' multibyte functionality behave the same way in 1.8.7 and 1.9.x.