similar to: Under which project , auto-dial feature comes

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.

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

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

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 Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet. It works fine to my local A@H box. I've tried... many things. I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working. Any Sipura experts out there? Eric.

Hi all, anyone could pls explain me what does it means ? It a part of zaptel.conf file. LBO= Line Build Out 0: 0 dB (CSU) / 0 - 133 feet (DSX-1) 1: 133 - 266 feet (DSX-1) 2: 266 - 299 feet (DSX-1) 3: 399 - 533 feet (DSX-1) 4: 533 - 655 feet (DSX-1) 5: -7.5 dB (CSU) 6: -15 dB (CSU) 7: -22.5 dB (CSU) Thanks Giordano -- No virus found in this outgoing message. Checked by AVG

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.

Is there a way to get asterisk to send you a email when it looses or an extension doesn?t re-register Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net Billing Questions: billing at upperclassman.net Rental Questions: rentals at upperclassman.net Maintenance: help at

Hi there i have some problems with some clients of my asterisk box, i have some cases when a client tried to make a call and there is no ring back only a silence and then the call hung up. I dont know why this is happening. I have the following stable asterisk version: CVS-v1-0-01/18/05-19:49:31 I did the an update a few days ago, the version that i had installed before was:

I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail(${EXTEN}@default,u|j) exten => _XX,3,Hangup() exten => _XX,102,Goto(110) exten => _XX,103,Playback(pbx-invalid) exten => _XX,104,Hangup() exten => _XX,110,VoiceMail(${EXTEN}@default,b|j) exten => _XX,111,Hangup() exten =>

Open the console with verbose turned up. Make a test call and see where it is hanging. That will isolate the problem. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Giorgio Incantalupo > Sent: Thursday, April 27, 2006 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial

Hi List; What configuration needed to let my FXS send and receive FAX? Regards Bilal

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