similar to: TDM800P FXO problem incomming call

Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks ----PGP Signature-- Satish

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130309/d82ddd47/attachment.htm>

I'm new to Asterisk and inherited this project so I apologize if this question has been asked a hundred time before. I did start with Google but I may not be asking the right questions, because I wasn't finding any answers. I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium TDM800P to interface with our six analog phone lines from the telco. Currently I have a single trunk

Dear all I have asterisk every time my Agent login in queue and useing queue but i want to staticly map that agent in queue so how do it possible and what configuration required for it ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Looking for last minute shopping deals? Find them fast with Yahoo!

Hi All, I have a server running Fedora 14, kernel 2.6.31.14, Asterisk 1.6.2.17.2 and Dahdi 2.4.1. I have the wctdm24xxp+ loaded with a Wildcard TDM800P with 8 FXO ports When a call is placed extension to extension, there is no problem... When an extension is used to dial out, the called number answers, but the server only detects the answer about 3 seconds later .. The outgoing

Dear all I have configure Asterisk 1.4 with sccp and configure Cisco phone 7975G model with Asterisk and it is working fine but i have one problem when i going to confance like when i call to someone and put it on hold and want to take in another New call to new person and want to take both live call in confrance how to do it in asterisk is there any configuration XML

Dear all I have small lan and i have configure hardphone with my asterisk with one E1 PSTN line now i have configue to use canreinvite=yes in sip.conf If i user conreinvite=no then my RTP goes throgh asterisk means asterisk come in media path and if i user conreinvite=yes then RTP path would be sip phone to sip phone ??? My all phone in LAN not behind the NAT so guessest

Dear all anybody have idea of this 2 card and performance vise which one is suggestable ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. -------------- next part -------------- An HTML attachment was scrubbed... URL:

What is SIP jitter buffer how can i test it ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Get easy, one-click access to your favorites. Make Yahoo! your homepage. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

I just installed a TE820 octal span T1 card, and it's not showing up in dahdi_hardware output. This was installed into a test machine that already has a TDM800P card in it, and that one is showing up and working fine. Is there some kernel module that I'm missing? Lspci: 05:04.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) 21:08.0 Communication

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

Hi , I am looking for squid-2.6 RPM for CentOS 4.5. I googled. But, No squid 2.6rpm was found yet. R u rinning squid 2.6 RPM on CentOS 4.x? if so pls let me know. I downloaded this squid-2.6.STABLE6-4.el5 .src.rpm from below link. http://mirror.centos.org/centos/5/updates/SRPMS/ Is it okay to build and installed on PRODUCTION servers? Hope to hear from you? -- Thank you Indunil

This is the message returned when I tried rinning it following a clean install of Wine a couple of days ago: $ env WINEPREFIX="/home/alan/.wine" wine "C:\Program Files\Sa ge\Instant Accounts\SAGE.EXE" fixme:system:SystemParametersInfoW Unimplemented action: 4132 (SPI_GETDROPSHADOW) Any known cure?

Hello, This is not a question directly related to asterisk. I am currently rinning ansterisk on a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently, it registered only 900MB. Can anyone tell me why thi is and a solution to this?? My Debian version

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc