similar to: international caller id support

Hi, I've purchased two x100p clones, and when I try accessing a line from asterisk with something like this: exten => _1NXXNXXXXXX,1,Dial(Zap/2/{$EXTEN}) (is that only supposed to put you on channel 2 or actually dial the # for you?) but I first hear noise, then a dial tone, but as soon as I start dialing numbers I get feedback and noise, and the call doesn't go through. Any

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

Hi, Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and

Hi, in response to a previous posting regarding getting the x100p to work, I was told to run "zap show channels," but when i do i get "no such command 'zap'" There was a previous posting on this, but the guy never posted the solution. thanks, Imran

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

Ming-Wei Shih wrote (Re: [Asterisk-Users] Best Linux for Asterisk) > I am running * CVS head on Gentoo/i586 > and Gentoo/Sparc64 (US60 2x450/1GB RAM), > they are running great. > > On sparc64 * does not compile out-of-the-box, > some hackings in the Makefiles are needed. Great stuff. Please, can you share your adjustments to the Makefiles with the community?! If you don't

Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: installed asterisk: make clean make linux 26 (for fedora core 2) make install installed zaptel: make clean make make install did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device 0000:00:0f.0 wcfxo: daa mode is 'FCC' found a

Hi I had asked for some help with the Asterisk Assistants http://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX and many have offered assistance with translations which I am grateful for and like to say thank you again. However, there hasn't been a single response from a graphic designer to offer help with a custom icon. Are there any graphic designers on this list at

Is there a way to override the SIP From Header that is used in the extension.conf Dial command? The default is 'asterisk@host'. I do not want to configure SIP accounts in sip.conf, but instead generate the SIP From-User within extensions.conf from data the user has entered interactively. Any idea? Henrik -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm trying to dial out on a vonage line connected to a zap channel using stuff like: exten => 200,1,Dial(Zap/2/${EXTEN}) but it doesn't work - when i dial in the extension, i can see on a phone connected to the same line that it's gone active - but there's no dialtone. also tried adding a wait period before accessing the line and exten =>

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

Hi, I've got a data set with a control group and a number of experimental groups, that have unequal sample sizes, and am measuring the number of people in each that respond yes or no. I'd like to use a dunnett test in R, where the syntax is supposed to be like: library(multcomp) test.dunnett=glht(anova_results,linfct=mcp(method="Dunnett")) confint(test.dunnett)

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Hi, I just setup two voicemail boxes - but after they record the wav file, they don't seem to email it even though i have the correct format in my voicemail.conf file. Aterisk doesn't give me any feedback besides the recording of the voip file... and then stays there. Secondly, I can't get the Playback() function to work. Thanks, Imran

Hi, the problems i previously had with "zap show channels" seems to be from an incorrect zaptel installation which is why I don't have a chan_zap.so file. I compile and do a "make clean, make, make install" of zaptel and do my modprobe's, and I was told to reinstall asterisk after that. I do so however, but it makes no difference. any hints? thanks Imran

Hi there, I have an X-Lite phone on my box and I'm trying to register it with a remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I know it's a pain to do because of SIP not working well with NATs, but I know there are ways to do such a thing...moving the Asterisk box outside the NAT is not a possibility at the moment. One thing we tried was setting up a VPN, but