similar to: IVR : Can't hear my message

I cant figure it out but why dont you dont you make an extension that you can dial and record yoursef. Exten => 100,1,Wait(2) Exten => 100,2,Record(FileName:gsm) Exten => 100,3,Wait(2) Exten => 100,4,Playback(FileName) Exten => 100,5,Hangup Antoine LOUIS <antoine.louis@gmail.com> wrote: Hello, I've reccorded a voice message for the IVR. (.wav, 16 bits,

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home at my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for most / 90+% of

Hi, I am in a predicament and any help/pointers would be appreciated. We are using chanspy to listen in on conversations. We are doing this via a web interface. The web interface lists all the ongoing calls. We click on a call and then my local phone rings allowing me to spy on the session I clicked on. But "most" of the time, when I start listening in, the two parties that are in

Adding an Answer() before MusicOnHold made it works. Thanks for everyone that helped ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101024/4c5546b0/attachment.htm

Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir:

I configured "parkcalls" and can see on "cli" Playing "digits/7" etc but I cannot hear them in the phone. I have: features.conf. [general] parkext => 700 parkpos => 701-720 extensions.conf: [extensions] include => parkedcalls What did I miss? -- #Joseph

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

Hey, I am a new to asterisk and softphones. Ihave recently installed and configured linux and 2 xlite clients all in linux fedora core 6. I have also made a dial plan for the two users. But when i dial from one xlite client to another i can hear the ring tone but when i answer the call i can not hear any sound. I have checked my microphone and its working fine. Please could anyone help me on

Hi, It worked finally with GSM Codec only enabled at client side.. Initially with G.711 (u-low) , G.711 (A-low) and GSM it didn't work. All enabled by setting [CLI] sip set debug on I saw asterisk having following logs.. -- Remotely bridging SIP/macbook-00000041 and SIP/tharindu-00000042 set_destination: Parsing <sip:tharindu at 192.168.1.3<sip%3Atharindu at 192.168.1.3> :

Hi, Anyone have any experience with these ??? http://cgi.ebay.com/UPS-1000VA-Brand-New_W0QQitemZ280141461506QQihZ018QQcategoryZ99265QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Thanks, Tuc

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EXTREME RED HOT ALERT! Buy Wednesday September 12th 2007 WW Energy Inc Symbol : WWNG This will EXPLODED TODAY, and will again TOMORROW! Yesterday we told you this stock would sky rocket, and today it did! Moving over 4.1 million shares, and up 100%. The stock doubled, and is going to double again tomorrow! There is an odour in the air a huge press release is coming out tomorroW! Current: $.02

Thanks for the links, I'll read them on my lunch break. I haven't heard much about whats coming.

Hi All, I connect SIP phones and H323 phones, with *. But, if I call from H323 to SIP phone, i can hear nothing. I'm using h323 library. I found some e-mails at the Digium's Mail-list with the same problem. But, I coudn?t find the solution yet. Could you please help me?, any idea?. It's quite urgent. Thanks. -------------- next part -------------- An HTML attachment was

So I've been fighting to get the X100P working. A battle which I've kinda won but not without a cost. Before I won the Zaptel battle I was able to hear all of the messages that asterisk plays. For example, when I'm accessing VoiceMail I would have been requested to input my password. This did work but now it doesn't. Asterisk does show that it is playing the file but no audio is

Hi, I'm testind Diax. I have flashing note about 1 new voice message. Can I hear it somehow from Diax gui, or must I call pbx to get message ? Thanks, Robert.

Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working before. Can anyone please help before my boss shoots me? JC zaptel.conf