similar to: 2 PRI on asterisk

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 have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website

Hi, I have a cisco sdsl modem to connect to internet via eth1 (192.168.1.2) local is eth0 (192.168.2.254) default gw is 192.168.1.1 the cisco forwards all incoming ports to 192.168.1.2. I connect from outside on port 24562, login is successfull, the ftpserver gives back the external Ip of the cisco as pasv IP to the client (its a setting in the ftpserver). It gives an ip from the pasv range I

Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Regards --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part

Hello, I've been using Samba on a Sun server but we recently discontinued using the Sun and switched to using Samba on a RH linux server. I can't get file sharing to work on the new server. When I test the connection to the samba server (velar) by running smbclient //velar/homes -U eric I get an error message referring to NT_STATUS_BAD_NETWORK_NAME. I can't find any reason for this

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Dear all I have one more question about codec what codec i use for LAN setup G.729 or Alaw which is best for LAN setup caz some people told me G.729 is use for wan link not for lan caz it is cost effective so can anyone suggest me best codec for asterisk and SIP phone Rgds satish patel --------------------------------- Don't pick lemons. See all the new 2007 cars at

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

We are running a 2-node cluster with Debian Squeeze (2.6.32-5-xen-amd64) and Xen 4.0.1 on the dom0 and Ubuntu lucid domUs. We are now using fully virtualized (HVM) vms. In our configuration the dom0 administers to domUs created and run on a SAN box accessed via private network (192.168...). A sample configuration file is pasted below. We have had little or no IO problem with HVMs created and

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

Dear all I have asterisk 1.2 and now i want to install E1 card with support Q.SIG singaling so which E1 card is best for my setup i need single port E1/PRI card which support Q.SIG Regards Satish patel --------------------------------- Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. -------------- next

Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was

Hi, I would like to produce a plot with a symbol on every nth point in a time series data, like the one in the following: http://www.phon.ucl.ac.uk/home/yi/ProsodyPro/EnglishFocus.png x <- seq(-100,1000,25) plot(x,type="l") Could someone help me out with the above example? Thanks.... [[alternative HTML version deleted]]

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 .....

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is busy -- Hungup

Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]-------[Mediant2k]------------[Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>