search for: echotest

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 'demo-echotest'...

...m. 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 => 600,3,Echo exten => 600,4,Playback(demo-echodone) exten => 600,5,Hangup Console shows something like that when I call: -- Executing Answer("SIP/206-081a7160", "") in new stack -- Executing Playback("SIP/206-081a7160", "demo-echotest&qu...

...xxxx> [Digital_out] type=peer secret=xxxxxx username=XXXXXXXXXX host=plasma.digitalvoice.ca fromuser=XXXXXXXXXX fromdomain=plasma.digitalvoice.ca insecure=very context = incoming_calls qualify=yes nat=yes EXTENSIONS.CONF [default] exten => s,1,Answer( ) exten => s,2,Playback(demo-echotest) exten => s,3,Hangup( ) [internal] exten => _NXXNXXXXXX,1,dial(SIP/${EXTEN}@Digital_out,30) exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN}@plasma.digitalvoice.ca,30) exten => 100,1,Playback(demo-echotest) exten => 611,1,Echo( ) [incoming_calls] exten => s,1,Answer( ) exten => s,2...

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:3...

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.

...Executing [*43 at from-internal:1] Answer("SIP/1000-08939150", "") in new stack -- Executing [*43 at from-internal:2] Wait("SIP/1000-08939150", "1") in new stack -- Executing [*43 at from-internal:3] Playback("SIP/1000-08939150", "demo-echotest") in new stack -- <SIP/1000-08939150> Playing 'demo-echotest' (language 'en') == Spawn extension (from-internal, *43, 3) exited non-zero on 'SIP/1000-08939150' -- Executing [h at from-internal:1] Macro("SIP/1000-08939150", "hangupcall&quot...

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

...le of beeps and then call hung ups. -- Goto (echo,s,1) WARNING[1227879616]: File pbx.c, Line 1160 (pbx_extension_helper): No application ' Background ' for extension (echo, s, 1) extension.conf has following lines under "echo" context [echo] exten => s, 1, Background (demo-echotest) exten => s, 2, Echo exten => s, 3, Background (demo-echodone) exten => s, 4, Goto(mainmenu,s,6) Could you please tell me where I could be wrong ? Regards, Tony

...see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and finally -- Executing Playback("Zap/65-1", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') at which point Allison's 'sultry' voice announces 'You are about to enter an echo test...' Can I remove this 3 second pause? It's rather annoying, and it doesn't happen when I dial out...

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

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

...non-zero on 'SIP/621-673f' [Oct 16 00:52:30] -- Executing Hangup("SIP/621-673f", "") in new stack [Oct 16 00:52:30] == Spawn extension (default, h, 1) exited non-zero on 'SIP/621-673f' [Oct 16 00:53:06] -- Executing Playback("SIP/621-88e8", "demo-echotest") in new stack [Oct 16 00:53:06] -- Playing 'demo-echotest' (language 'en') [Oct 16 00:53:26] -- Executing Echo("SIP/621-88e8", "") in new stack [Oct 16 00:53:33] == Spawn extension (default, 690, 2) exited non-zero on 'SIP/621-88e8' [Oct 16 00:53:...

Hi all, DIAX 0.9.9g is available for download (including the updated help file and web page) from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.9g (from 0.9.9f): - during a call, accept DTMF tones as monitored events to trigger output commands - call timer on the phone display - Swedish language added - can run a command from the

...;96' (AUTHENTICATED) at 212.202.169.118:4569 -- Accepting AUTHENTICATED call from 212.202.169.118, requested format = 1024, actual format = 1024 -- Executing Macro("IAX2/96@96/2", "echo") in new stack -- Executing Playback("IAX2/96@96/2", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'de') == Spawn extension (macro-echo, s, 1) exited non-zero on 'IAX2/96@96/2' in macro 'echo' == Spawn extension (imatris, 600, 1) exited non-zero on 'IAX2/96@96/2' -- Executing Goto(&q...

...ther provider which put our server on a dmz, so that now we have our server with private ip but reachable from the outside via port forwarding on a public ip. Now every communication with asterisk is mute, calls are relayed by ser, connections estabilished, but no voice either with sip or with demo-echotest (* log says he is playing echotest but I can't hear anything!). I thought this was a dmz firewall + rtp problem but ports in rtp.conf are open with forwarding (udp). This is current network situation: myserver: private ip 10.0.0.229, ser running on port 5060, asterisk on 5061 (sip), rtp ports...

...and off every codec and still no luck. Asterisk says it's playing the sound file but I just don't hear anything. I can't find any reason for this. I've tried the latest tar and CVS with the same result. sip*CLI> -- Executing Playback("SIP/hal-ac2d", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') Feb 1 05:13:57 NOTICE[12285]: rtp.c:317 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible sip*CLI> the RFC3389 message doesn't appear until after a few seconds of silence, I'...

...and I used a different install prefix for *all* forementioned packages. (all the same, of course) The thing is, soon as I get SJPhone to connect to asterisk (which happens fine), I tried the echo service by dialing 600. Then I get: -- Executing Playback("SIP/julio-5ae7", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') Jul 4 14:35:42 WARNING[13521]: file.c:550 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 600, 1) exited non-zero on 'SIP/julio-5ae7' Then I dialed 1000, for the "congrats&...

...samples". Then I proceeded with installing asterisk-addons and asterisk-sounds as suggested. The thing is, soon as I get SJPhone to connect to asterisk (which happens fine), I tried the echo service by dialing 600. Then I get: -- Executing Playback("SIP/julio-5ae7", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') Jul 4 14:35:42 WARNING[13521]: file.c:550 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 600, 1) exited non-zero on 'SIP/julio-5ae7' On my SJPhone I can see the sound is being...

...be going wrong? As a test case, I call my echo() extension from the remote phone. The connection works fine but when I hangup the phone and get information from the Asterisk console here's what I see: [Jan 27 10:27:00] -- Executing Playback("SIP/scottbhome-f4de", "demo-echotest") in new stack [Jan 27 10:27:00] -- Playing 'demo-echotest' (language 'en') [Jan 27 10:27:19] -- Executing Echo("SIP/scottbhome-f4de", "") in new stack **** I hangup the phone here **** pbx*CLI> show channels Channel Location...

....org/wiki/view/Asterisk+SRTP updated test srtp server (asterisk SVN-trunk-r61760 + latest SRTP patch) voice2.fpf.slu.cz test sip accounts 700:700 701:701 702:702 extensions.conf exten => 600,1,Set(_SIPSRTP=optional) exten => 600,n,Set(_SIPSRTP_CRYPTO=enable) exten => 600,n,Playback(demo-echotest) ; Let them know what's going on exten => 600,n,Echo ; Do the echo test exten => 600,n,Playback(demo-echodone) ; Let them know it's over exten => 600,n,hangup exten => 610,1,Set(_SIPSRTP=require) exten => 610,n,Set(_SIPSRTP_MIKEY=enable) exten => 610,n,Playback(demo-echot...