similar to: Configuring behaviour of flash hook

Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Why can this happen? The host stanzas in iax.conf have raw IP's, so no DNS monkey business here.. An inquiring mind wants to know.

I'd like to set up a "silent attendant". By this I mean when someone calls me I'd like for them to hear the comfort ringing tones, but for the first 5 seconds I'd like to give them the option of pressing 9 to send the call to an alternate extension; if they don't press 9, the call goes to a default extension. For most callers I just want standard PSTN behaviour, only a

Hi: I upgraded SuSE to 10 and Asterisk to trunk and now after deleting all modules and previously compiled stuff and recompiling asterisk, zaptel, and libpri, I get this failure of asterisk to start: [pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style pbx_realtime.so (0x31) loaded RTLD_LOCAL => (Realtime Switch) [chan_mgcp.so]Apr 25 03:36:41

Hi, When I setup a user, I give them an extension like 570xxxxxxx. This is fine and dandy while in one area code, but we've since gone to other area codes. I'd like the user's to retain the ability to dial 7 digits no matter what number they have. Any thoughts on how to do that? EXAMPLE: User has number 7175551212. I want that when they dial 3235555 it dials 717-323-5555.

I think A@Home allows a user to search a directory by either first OR last name, right? I don't know for sure since I don't run A@Home. I would like to offer that functionality in my system - and I'd have done it by now if there was a prompt where Allison asks "press 1 to search by first name, press 2 to search by last name". But I don't think that prompt exists. Can

Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next

Is there a way to have an extension match on a sip address? I've tried the obvious - _.@. but it seems to behave just like _. which is no good. Is there a better way? -- Jon-o Addleman - http://redowl.dyndns.org

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

Hi After migrating from 1.2.4 to 1.2.5 I noticed that: show application dial does not show the 'R' option anymore. Has this become an undocumented feature or has it gone completely? Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00

Hello all, is it possible to make an outgoing call transferable for the dialing phones like the 't' or "T" option on the Dial-Command does this for incoming calls? Does someone have any idea? Thanks Hans-Peter Straub -- -------------------------------* I-NetPartner GmbH Hans-Peter Straub Seewiesenstrasse 12 D-73054 Eislingen -- Phone: +49 7161 9849955 Fax: +49 7161

Hi, With a digium dual PRI card (dual span). Is there any reason I can't have both PRIs being PRIMARY timing sources? They are both from different CLECs, and as such I need them both to do their own timing.

I'm trying to get asterisk to pass through a call without requiring supervision on the line. Any thoughts? Thanks, Frank

Hi, I have a queue that plays music when a call comes in. To be able to do that I need to Answer() the call first. After a timeout in this scenario the call should be transfered to an extension using a GoTo statement to the extensions context. The problem is that as soon as asterisk Answers the call it can not play a ringtone (or other tones) back to the original caller when executing a Dial

I get MOH when the caller hangs up. Is there any way I can just get Busy tone. Regards Michael Knill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060522/4905a84b/attachment.htm

Hi, Is there anyway in Asterisk to know that outgoing call has been forwarded to voicemail by the callee system? Some of my users don't want to connect the call if its forwarded to callee voicemail, so I am wondering if theres anyway to identify this in Asterisk and drop the call. Thanks Nitin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with kewlstart signalling. When an outside caller calls the server, the TDM-400 goes off-hook and provides a ringing tone to the caller. If the caller hangs up before the receiving party answers the phone, Asterisk fails to detect the hang-up. The TDM-400 stays off-hook, hogging the line, while Asterisk rings the

I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below: Jun 14 09:38:12 WARNING[18292]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/3004-fcfb(256) to SIP/3003-c1c3(2) Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make SIP/3004-fcfb compatible with SIP/3003-c1c3 ==

Hi all, I've got a fairly simple setup, a 4 port zaptel T1 card with 1 PRI and 2 flat e&m T1's coming into it, and its working perfectly except for one small thing. When people dial any of the numbers on either the PRI or the T1 from outside of the pbx, they dont get the "ringing" sound. It just directly connects the call. I know this isnt much, but its started to