search for: 2125551212

...y a custom unavailable greeting. When I record the greeting (using the std voicemail method), it creates a valid .WAV file, which I can play using the Playback command. However, when an incoming voicemail is received, Asterisk plays the default system greeting. I see this warning in the logs: //...2125551212 is the sample voicemail box in this example...// [Nov 11 12:30:00] WARNING[8343] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/consumer/2125551212/unavail/msg-001.WAV: No such file or directory I'm not sure if this is actually bad though, because it is recording the mess...

...s/2005-December/132327.html I've also tried moving the dtmfmode=rfc2833 to the general context in sip.conf. Here's the current sip.conf configuration: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no ;dtmfmode=rfc2833 register => 2125551212 at sip.broadvoice.com:lePassword:2125551212 at sip.broadvoice.com externip=199.171.51.236 localnet=10.91.180.36/255.255.255.0 nat=yes [authentication] [1000] type=friend context=phones host=dynamic [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com f...

Just setup *, got a developers kit FXO where the incoming/outgoing pstn is plugged in. I've then got an IAXy that is plugged into a Philips DECT phone. * is setup so that the [bell] section rings the phone - exten => s,1,Dial(IAX2/myuser,30) What's happening when someone calls my number is that the phone rings 3 times before you hear the dect phone ringing. Anybody got any ideas as

Guys, How would you block inbound call's? for example person who is calling me is 212-555-1212, and I would like to do not receive the calls from this person and give them busy tone. What should I write in asterisk config files? and in to witch file should I write it??? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

I have a situation, where call comes in on one PSTN line, (say 2125551212), and then I want to change the Caller ID & ANI to 2125551213, and then forward the call to 2125551214. [incoming] exten =>2125551212,1,Goto(outgoing,s,1) [outgoing] exten => s,1,Wait(1) exten => s,2,Answer exten => s,3,SetCallerID(2125551213|a) exten => s,4,wait(1) exten =&...

Hello, Im trying to modify the 'From' field in my sip headers in order to include extra info (user=tel) as it follows: >From :"" <sip:005114824403 at 200.91.0.146;user =tel> However asterisk is still doing this header: "11111111" <sip:11111111111 at 1.1.1.1>;tag=as167b4b82 Is there a way to accomplish this? Ive been also looking up AGI but without

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm

...and wondering if someone can spot a problem in the logs, below. Some details to help read this log (I've changed these numbers for privacy purposes): . My Asterisk server is behind a firewall. It's internal address is 192.168.71.1. . My public IP address is 123.123.123.123 . I am calling 2125551212 . My Broadvoice phone number is 9145551212 Here is the log: == Using SIP RTP CoS mark 5 -- Executing [912125551234 at dialplan-siegel:1] Macro("SIP/siegeld-00e08e00", "dial-sip,12125551234 at sip.\ broadvoice.com") in new stack -- Executing [s at macro-dial-sip:1] Di...

Can any one suggest a little code to either ring a cell phone when a new VM message is recorded, or send a text message? Basically outside sales people want to know they have a new message, but don't want to be interrupted to take a forwarded call. While a message by message notice would be nice, even just a single notice on the first message would be an alert to call for messages.