similar to: FXO incomming call hangup problem

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 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not .....

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!

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

How I change my configuration to reduce this issue. I have this setting on my zapata.conf signalling=fxs_ks group=1 callgroup=1 pickupgroup=1 channel=1 signalling=fxs_ks group=2 callgroup=1 pickupgroup=1 channel=2; singalling=fxs_ks group=3 callgroup=1 pickupgroup=1 channel=3; singalling=fxs_ks group=4 callgroup=1 pickupgroup=1 channel=4 and for outbound calls I have this context on my

I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot

Hi, Here is the configuration of the server that I currently have extension 100 (SIP) =>(SIP)asterisk server 1.8.18(IAX trunk) <===>(IAX trunk)asterisk server 1.4.32(SIP) ===> SIP Providers The issue is while dialing out from extension 100(sip) if the providers sends back 180 Rining the SIP extension(100) won't hear the ringback tone, where as if the providers send 183

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer ; Answer the

Dear all I have special requirement of samba domain security...i want to bind user with machine so that use only ...and only able to login with that same machine ...means user can not login in to any other PC or machine only access on own machine...is it possible with ldap attirbutes ..? $ cat ~/satish/url.txt

Dear all I have smb+ ldap setup not everything is fine but i want to assign some right to perticuler Group so they can change TCP/IP properties and change system time and do some other right Is it possible to give some privilages to normal users ??? $ cat ~/satish/url.txt http://www.linuxbug.org

Hi, there could you help me coding this problme? I am just starting to leard the R. So I really need help Question is from Ross, Simulation, 4th Edition. ch3 14. with x1=23, x2=66 Xn=3*Xn-1+5*Xn-2 mod(100) n>=3 we will call the sequence Un=Xn/100 n>=1 find the first 14 values thank you sophia

I am running * with a Junghanns quadbri that should allow us to integrate our ISDN house telephone system with VOIP. Preferably I would like to run a setup, so that our internal ISDN phones on an S bus are not aware that * is sitting in between. With the configuration below I run into the following problems: 1. On outbound calls, I get the normal rining call progress tone althought the

Well I dont want it to operate as a group. I understand that it is *8 on most PBX systems but I would like it to work as dialing *8+extension-thats-ringing to have it go over to my extension. I am having a lot of trouble finding more information from the wiki and google. Thanks! -James -----Original Message----- From: asterisk-users-bounces@lists.digium.com

created you must place it in your web directory on the server. =20 I chained the command and also wrote the output to an xml file in the web directory. The command looks like this: =20 'php /etc/asterisk/directory.php.txt > /var/www/html/directory.xml' =20 System Speeddials using Services Button =20 =20 For speed dials I modified the php code to look to a specific file in the