similar to: Network Voip Carrier Termination (Off Topic)

Dear listener I'm having problem trying to setup a 640x480 image background with latest syslinux... the system start correctly but graphics is not shown at all... only text is visible... are there minimum requirements for a graphic extlinux background ? CPU is a Vortex86Sx (with no math processor) at 300MHz, 128MB RAM video card is a XGI Volari Z9s with 32MB RAM is there a way to

Dear List i have upgrade my * box with the latest CVS version of Asterisk Stable 1.0 and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems work well for now but i have a little amount of traffic (25 IN/OUT calls) i only notice this Warning.. What kind of error is? ------------------------------- Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for

Dear List I have notice this kind of problem between my two * box. My scenario is : Iax GSM IaxClient----->PBX1------------>PBX2-->TDM today CVS Stable V1 I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join the two call i can see the log below from my PBX1, i can speak for

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

I'm confused about VoIP termination service. Would this mean that I could have an Asterisk server in my basement and receive/call PSTN phones via a VoIP termination provider? The "call" out from my box would go over the Internet to a termination provider which would in turn convert it to analog over the PSTN to someone's plain old phone? And vise-versa, a plain old phone caller

Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/5068aaab/attachment.htm

Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks!

If you connect an * system to a company like VoicePulse with there Connect service you are able to get incoming calls unlimited for 7.99/mo and outgoing at .0295/minute. VoicePulse also offers an unlimited business plan. Are you able to sign up with VoicePulse unlimited plan and connect an * system to it? Also, with VoicePulse connect you're able to make unlimited simultaneous calls

I realize this is a shot in the dark, but I'm trying to find a VoIP provider that offers 402 or 712 area code DID numbers. I'm almost completely convinced that no one offers these area codes (eastern Nebraska, western Iowa), however considering the wide audience of this mailing list I thought this would be a good place to ask. I would prefer a provider that allows for Asterisk use, but I

after rebooting I get very often: ... mfi0: COMMAND 0xffffff800132d990 TIMEOUT AFTER 659 SECONDS mfi0: COMMAND 0xffffff800132d990 TIMEOUT AFTER 689 SECONDS mfi0: COMMAND 0xffffff800132d990 TIMEOUT AFTER 719 SECONDS ... another reboot usualy fixes this. danny

Hey guys, I'm actively trying to get the "big" picture on how all this works and relates to each other. I've gone through some basic examples from the book and from the sample files just fine. Now, I've setup an account with a VOIP provider which does IAX termination (exgn.net) After getting an account and following their steps, I can make calls out using my IAX (cubix) and

Hi all. I'm looking to change from a standard telephone line to a VoIP phone line at home. I'm looking for recommendations for VoIP providers that I can use with Asterisk. One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the

Is there a recommended partitioning for dom0? Here''s what I have planned. /swap 2GB /boot 100MB (ext3) / 5GB (ext3) xenmachines LVM volume group for the remaining diskspace dom0 will install in 5GB / and each domU will have a swap and root logical volume in "xenmachines" Thank you, _______________________________________________ Xen-users mailing list

Hello all, ( I apologize if this posts twice ) Here is my situation. I have 3 buildings linked with 100mbit fiber optics (2 runs that come to the corporate office). I have 3 RH9 boxes, one at each location. Each box at the remote locations have 4 NICs, one for the fiber link, one for LAN, one for the VOIP box and one for the internet connection. The corporate office has 4 NICs also, 1

Hi, The second CentOS user interaction Dojo is taking place at Phoenix, AZ, USA on the 10th May 2013. And once again, we have a great line up of speakers covering a broad spectrum of technologies that people running CentOS usualy care about most. For details on the speakers, the topics and the venue : http://wiki.centos.org/Events/Dojo/Phoenix2013 The early bird ticket sales end on the 30th

Does someone understand this? I do not. Niels. ----- Forwarded message from laurent Protois <protois at ensea.fr> ----- Subject: NVIDIA and Privilege Separation From: laurent Protois <protois at ensea.fr> To: provos at citi.umich.edu X-Mailer: Ximian Evolution 1.0.7-1mdk Date: 10 Jul 2002 09:29:45 +0200 Hi Niels, i have a little problem with openssh 3.4 and Nvidia kernel driver:

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

Dear all; I have and asterisk box receive a phone call from a VoIP carrier and pass it to internal SIP clients, it worked fine when using g711, when it comes to g729 call established successfully and there is some rtp flows but dead air on both side, any ideas? Regards

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm running stock FreeBSD with services running: samba (connections allowed only from local network), lpd (same), bind (all interfaces), apache (all), zope (local) This machine is home gateway/http/printserver. Recently some strange things happened as my printer all of sudden started to print stuff when nobody prints... luckily (or

Hellos, My voip carrier has assigned me a IP based account...where they only give me the IP to call through. I have setup the dial plan exten => _7XXX.,1,Answer() exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) exten => _7XXX.,4,Hungup() Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc