similar to: Questions About Fax for Asterisk

(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not

I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I'm new to Asterisk I can't get the error why this is not working. To me it all

So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry

Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can?t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no

Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to send to email the voicemails. i would like to get rid of the analog fax machine and use asterisk to send/receive faxes. I do have a PSTN line with a SPA3102 adapter to interface it to asterisk. The number of the PSTN line is dedicated to faxing only. So i would like to: -receive faxes to asterisk and then send it as PDFs

Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On the Python Tutor mailing list Kent Johnson uses a script to find the top posters for the year. If this or something like it has been posted, sorry for the noise; 2008 ==== Steve Totaro 796 Tzafrir Cohen 749 Tilghman Lesher 496 Alex Balashov 354 Olivier 334 Philipp Kempgen 251 Gordon Henderson 242 Atis Lezdins 239 Jay R. Ashworth 230 Doug Lytle 207

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I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get "Ring requested on unconfigured channel". How can I busyout these these channels so that incoming

It would be cool to get one of these and see if it can be hacked and loaded with your favorite SIP or IAX softphone. Looking at the pic, it looks like the dongle is both a soundcard and memory stick. Heck, I would be glad to have it if I could get the soundcard to work. Might as well since it is free after rebate. http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem

Deal All Asterisk Expert If this possible to Create Voice Recording System Beside Main Asterisk PBX?, so Call be handle by 1 Server and Recording by other server. 1. How to accomplish. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090325/d57465cd/attachment.htm

Just been getting lots of failed SIP registrations to a system here. All coming from Taiwan. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy

Hello all, I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet? Many thanks, Christian

hot to asterisk multiport...??? example 5060, 5061, 5080 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070815/8a515309/attachment.htm

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

i am just curious how many * systems are in the real world with more than one user. do you run a certain version? you dont update CVS do you? any admins running a system of over twenty? over fifty? over one-hundred? i deal with 3com and nec systems all day (i am cerified in the 3com nbx advanced network telephony, elite voice mail, CCNA, A+, nec ipk, and soon to be "asterisk school of

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I