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I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's possible to do this? I've ever tried splitting 'peer' and 'user' part in

...The problem is that the call always goes through after I've entered the 9th digit. My service provider is BroadVoice, my phones are Grandstream GXP-200's. DTMF is set on the phone to be via SIP INFO and Early Dial is set to "Yes". I have this in my plan: [outbound-long-distance-forio] ; this is how outbound long distance calls are handled ; international numbers are max 15 digits included 2-3 digit country code (E-164 ITU protocol) ; we assume 9-15 digits exten => _011XXXXXXXXX., 1, Macro(forio-dial-outbound,default,${EXTEN}) ; long distance exten => _1NXXNXXXXXX, 1...

Hi, I'm provisioning an office with limited cabling. I'm looking for a desk based wifi phone. Most of the ones I've seen are handsets. Any ideas? Thanks, WILL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051007/1d2cc49a/attachment.htm

Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going (for example) to the voice mail on my cell phone. Scenario * Call comes in,

I downloaded asterisk-sounds-1.2.0-beta1, superused, then typed "make install". The installation stopped with the following error: No description for sounds/access-code.gsm make: *** [datafiles] Error 1 Does anyone have any useful tips? I'm running Debian 3.0. Thanks, WILL -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, We're having problems dialing out to Asterisk from our Grandstream GXP-200 phones. About 2 of 3 times, when we dial, nothing happens. Looking at the console in max debug mode, there are no messages except the following: Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 9913b47bcd7 aeb52@192.168.10.100 for seqno 4524 (Critical Response)

Hi, I'm having problems with an occasional disconnect from phone calls while my phone is on mute. This is a problem with long conference calls, for example. I've a GrandStream GXP-2000 and Asterisk 1.2.1. Anyone have experience with similar issues? Best, WILL