similar to: Asterisk E1 card support Q.SIG

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

Dear all I have setup with mediant 2000 with avaya now i want to install E1/PRI card with asterisk and trunk with E1 with Avaya E1 port so i want to buy E1 card for asterisk so which card is best and cast effective for my setup i want 1 port E1 card so can you suggest me which card is best for my setup and i want QSIG singaling with avaya Regards satish patel

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 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 install 2 port pri card on asterisk but i dont know how to incomming call goes in to IVR and how to route call outside base on pattern match means if some one call on mobile phone then use PRI 1 and if call on landline phon call route through pri 2 how to make dission base on pattern number Rgds satish patel

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

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 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 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

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

Hi All, I have the following section in my smb.conf file: [Satish' Hindi Alltime Hits - Gems] path = /mnt/d-drive/Music/Satish' Hindi Alltime Hits Gems browseable = yes public = yes readonly = yes guest ok = yes I am able to browse this share through win2k but not from winNT. But If I change the share

Hi, I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Thanks a lot in advance, best

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

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk

Hi, Is there an easy option in rsync which enables it to sync the data first and then the symlink pointed to it. For e.g. at destination I have a symlink "new" which points to anything latest at source. The problem is that when the sync happens it updates the "new" link first before the data it is pointing to has arrived. This causes me all sorts of problem. Please

Satish: Does your app uses any AX controls?. If so, it should be registered. Partha Saradhi -----Original Message----- From: Satish A. Lele [mailto:satishl@spidersystems.co.in] Sent: Friday, July 02, 2004 11:53 AM To: wine-users@winehq.org Subject: [Wine]Active X Component Can't create object error while running vb application Hello, I am using Red Hat Linux 9. I am trying to run

Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

The last post I saw regarding this was June 2003 (before my time with *, I started in September): http://lists.digium.com/pipermail/asterisk-users/2003-June/013324.html Is there any support for Q.SIG in * at all? The Norstar MICS uses a MCDN (Meridian Customer Defined Network) key to enable the "SL1" protocol which from everything I've been reading is just Q.SIG. This protocol

Hi! I want to test Asterisk<-->Siemens HiCom integration using Q.SIG instead of ISDN. I did not find any documentation about Asterisk und Q.SIG. Thus, I wonder is it sufficient to set "switchtype" from "euroisdn" to "qsig" or are there any other things which I have to take care of? Thanks Klaus