similar to: TDM11B Hang up detection not working in France ?

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, 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, 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, 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

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

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 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 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

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function

This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

I have googled this and come up empty. Has anyone had any problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am getting when I run make. app_nv_faxdetect.c: In function `nv_detectfax_exec': app_nv_faxdetect.c:210: error: structure has no member named `cid' app_nv_faxdetect.c:227: error: structure has no member named `cid' app_nv_faxdetect.c:265: error:

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 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