similar to: compiling zaptel-1.2.5

I'm trying to compile Zaptel driver with the ztdummy. I have no hardware cards from digium. I tried following steps: http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy I'm running : Linux version 2.4.27-2-386 (horms@tabatha.lab.ultramonkey.org) (gcc version 3.3.5 (Debian 1:3.3.5-13)) #1 Wed Aug 17

Hello to all, Is there an "how-to" for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards Harry PS: I use hylafax server. ___________________________________________________________________________ Faites de

Hello, I read http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPredirect I wish to configure asterisk as a redirect server. I have badly understood this command . ASTERISK | sip agents nated ======SER When sip agents send INVITE to the proxy i want ser send this one to ASTERISK . How asterisk can send 302 messages with the good info in

Hello, I patched asterisk patched with the latest t38 support . I would need some people for tests. Regards harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el.

Hello, I 've installed both cacti and res_snmp for monitoring. Does res_snmp is able to send snmp traps when hardware is out of service or others status ? Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos

Hello, I have an error when installing AMP, when I do ./install_amp --debug, it show me : Connecting to database..FAILED [DEBUG] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)] ** mysql://user:pass@localhost/asteriskamp Try running ./install_amp --username=user --password=pass (using your own user and pass) [FATAL] Cannot connect

hello, Is there somebody wit a h323 terminal ? ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el. Rendez-vous sur http://fr.yahoo.com/set

Hello every body. I have this PCI card : DM/V1200-4E1 spec in this site: http://www.intel.com/network/csp/products/3967web.htm Can i use it with Asterisk, is it compatible ? Thank you in advance. ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez

Hi does asterisk act as SIP proxy ?, like SER any documents if does, will be great help ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060516/9fce4aca/attachment.htm

In my attempt to setup a single FXS line I have been following the instructions for "Telephony Card Drivers" on the asteriskdocs.org site. I have managed to checkout, make and install the zaptel code and can load the zaptel module but when I attempt to load the wcfxs module it tells me: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

Hello, I look at the mobicents project. Somebody has experience within both projects ? Regards Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el.

Hi all I install sphinx2 successfully but when i execute sphinx2-server, i have the error below: ad_oss.c(105): Failed to open audio device(/dev/dsp): No such device FATAL_ERROR: "server.c", line 476: ad_open() failed What's the matter? I also want to know how i can do in Asterisk to use the sphinx server.Must i write entirely an eagi script? And when it's wrote, which

Thanks for these informations I would have prefer to receive them from asterisk-users instead of serusers !! May be they are sleeping . Ok i have not installed spandsp because of i don't find some scripts like in hylafax for mail2fax fax2mail i've just patched chan_sip.c Regards Harry --- Alexandr Dubovikov <shurik@start4.info> a ?crit : > On Wed, Apr 26, 2006 at 11:27:01AM

Hi asterisk, openser, ser users. I have to check video support between asterisk, open(ser) and rtpproxy . ASTERISK (b2bua+registrar server) | | | | SER + rtpproxy | | NAT | | sip agents (with video support) Both signalling and media channels are kept in the path of SER+rtpproxy and ASTERISK . I can

Hello, Is there Somebody to provide me a DID numder on a voip gateway which one support t.38 to test FOIP ? Regards Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en

Hello, I need to install a h323 channel in order to asterisk act as a sip/h323 translator . I want to use gnugk in full proxy mode for the h323 terminals nated . Which h323 channel for asterisk and gnugk h323 oh323 or ooh323c ? Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver

Hello, How can I park a call or put on hold a caller from an analogue to sip agents ? PSTN===FXO/asterisk=====sip agents When I press hold key or #800 the channel is hangup ?? Harry Regards ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez

I use res_snmp.so with asterisk do you provide mib --- David Yat Sin <davidy@sangoma.com> a ?crit : > Hi Harry, > The Sangoma Card when used for TDM Voice will work > under zaptel, so you > would need to perform the SNMP through Asterisk. > > Regards, > David Yat Sin > Sangoma Technologies > (905) 474 1990 x119 > (800) 388 2475 x199 > MSN:

Hello, Can we park a call or put on hold a caller in a queue ? I have sip polycom phone but when i press hold key or #800 i can't neither park call nor hold this call . Is it possible ? Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez