search for: voiceprompt

Hi, because many people contacted me about this the last couple of days and I guess most of them are on this list anyway: - Yes, our new German voiceprompts for Asterisk 1.4 are ready and can be downloaded at http://www.amooma.de/asterisk/service/deutsche- sprachprompts/ - Yes, we are in discussion with Digium about including them into the normal install process. I have no timeline but a GO from Digium. So it should be available pretty soon. F...

Hello ! Are there any plans at Digium to include also german voice prompts ? Thanks regards Hans

Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple "hello world" dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under

Hi, I am running asterisk 1.6.1.6 with a howler screamer card. I have g729 and alaw trunks from a pbx /sip providers. The howler screamer will only transcode from g729 to alaw / ulaw but my voice prompts are in SLIN and throws errors when i try and access these applications. Is it simply a case of converting the prompts into other codecs and asterisk will pick these up? ? Thanks

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

...a macro where the caller is given the opportunity to leave a voicemail message. It's to late to answer() the call in the macro, but I guess the voicemail()-application automatically anwers the call ?? How about an IVR-prompt and a queue ? Do I need to answer the incoming call before playing a voiceprompt and before sending it into a queue ?? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/dab3448d/attachment.htm

...the "vmusers" view mandatory for Asterisk to function properly? 2) in "my" world, sip-users have a language. At certain points, Asterisk will communicate with "my" users regarding voicemail. e.g.: a) Asterisk delivers the message left as a wav-attachment by email, b) VoicePrompts guide the user through the voicemail-menu So far, I am able to hard-code the language for the voiceprompts in extension.conf with Set(CHANNEL(language)=fr). But this set the language for everybody. Also, you can customise the email template ($emailbody, etc) for voicemail delivery in voicemail.co...

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network sniffing using ngrep and verified that the voicemail app is indeed not sending _any_ udp/rtp packets towards my sip fones. I did restore old, wor...

...m quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... n...

...a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot) and a few calls later i get the voiceprompt "All circuits are busy....." I know the PSTN line is on hook and ok to use. Looking in the log file i can see a line saying unable to create a ZAP channel. (See extract below) Sep 16 07:47:46 VERBOSE[2012]: -- Executing SetVar("SIP/200-361a", "DIAL_NUMBER=850220")...