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Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.

Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello, How many digium cards is supported per asterisk server ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello, Asterisk sent back 407 proxy authentication . How can avoid this ? I set insecure=very without success in sip.conf and my sql server . Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hello, I wish to set a sip uri sip:info@mydomain. I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten => info,1,Answer() exten => info,n,Dial(Sip/84,10) exten => info,n,Dial(Sip/85,10) exten => info,n,Hangup Ser forward sip:info@mydomain

Hi, I have some trouble with the Monitor() application. I start and stop it via the management interface, giving no special parameters except the channel name. What happens is: - if I specify WAV as the format, the resulting files are exactly 44 bytes big and contain nothing at all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards

The list of your interest is R-help not R-sig-mac stefano Il giorno 26/gen/06, alle ore 01:20, Sylvain Charlat ha scritto: > Hi, > > Is there any simple way to get histogram for different levels of > factor? > > Say you have the following data set: > > Island Sp.diam > Moorea 1.21 > Moorea 1.27 > Moorea 1.28 > Moorea 1.22 > Moorea 1.28 > Rurutu

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the

Hi, i need some help with Effect.Appear/Effect.Fade i would like fade in and out an element on hovering an element(-> tooltip behavior), but when i''m using Effect.Appear/Effect.Fade both effects will continue to play and overlap each other, resulting in unpredictable behavior. <code> <button onmouseover="Effect.Appear(''test_div2'')"

I had a problem with the send_file method. I saw that some of you had the same problem but with no real solution/explanation first problem the :stream options, and not ''streaming'' as I see in some doc or book. the second problem : when you set :stream to true, there''s a use of syswrite that should not be. I detailed this in response to a ticket opened by stuart

this game run perfectly, but i can't make the joystick working first, my specs : athlon-xp 2600+ , 512mb ddr , geforce fx5700 , gentoo linux my use flags for wine 0.9.5 : +X +alsa -arts -cups -debug -esd +gif +glut -jack +jpeg +lcms -ldap -nas +ncurses +opengl +oss -scanner +truetype +xml2 i have a cyborg 3d usb it's detected in linux, it works in native games, with the joydev module i

Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=====Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar server, pbx functions, ... SER/OPENSER look for domains in URI. if domains are handled by SER/OPENSER

Hi all I'm a newbie in asterisk.I install asterisk server successfully. I configure this server to traverse NAT. Using Xlite clients, i make a call between 2 local networks through Internet.Asterisk server is installed on a host with public IP. client A (in the LAn A) and client B (in the LAN B) are registered. When i make a call from the LAN A to the LAn B, everything goes well.But, when

Message: 6 Date: Fri, 03 Mar 2006 17:32:47 +0000 From: Conrad Wood <asterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.net> Subject: Re: [Asterisk-Users] Problem with NAT!!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem! Thanks in advance! bets regards Serge --------------------------------- Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international.T?l?chargez

Hi all, May I have to patch asterisk-1.2.x with this patch http://bugs.digium.com/bug_view_page.php?bug_id=0002859 to configure an outbound sip proxy in sip.conf ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et

Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Bests regards Serge ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et