similar to: Kernel 2.6 and zaptel data

We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest. ftp://ftp.digium.com/pub/asterisk Enjoy the code. Special thanks to all the bug marshals and contributers and to everyone who has supported Asterisk

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

Thanks Peter, Yes, indeed the problem seems to be exactly what you describe. It's overhere the same. If I dial a mobile number it disconnects immediately when I hangup the mobile. But for analog numbers it takes around 10 seconds or so... Well, at least now I know how to debug pri :-) Walter. On Thu, 29 Jul 2004, Walter Klomp wrote: > However, if I dial-in from the SIP phone to my

Hi *, I want start with a setup of Asterisk with a clean PC. This PC is a SMP-Machine with two 466MHz CPUs, a Acer ISDN card and a AVM Fritz! PCI card. Which Kernel is better for my constellation (Asterisk with SMP, CAPI and ZAPHFC)? Kernel 2.6.x or Kernel 2.4.x? Regards Bastian

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all. I'm new to this so please forgive my ignorance if I missed something obvious. I've set-up asterisk 0.9_1 on a FreeBSD 4.10 server (I know it's not linux but that's all we have available at that stage). After some struggle to understand how everything works, I set up some SIP accounts for test purposes. I can log in,

I am trying to compile chan_capi 3.3.4a, but I end up with lots of gibberish. Near the top it states that capi20.h doesn't exist. Searching for the file, several show up: # find / -name capi20.h -print /usr/src/linux-2.4.21-144/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/smp/include/config/isdn/capi/capi20.h

I replaced my X100P cards with two TDM04B fully populated (8 FXO modules). They are working fine, I can make and receive calls, but I noticed all modules are always in GREEN state, even if I disconnect the line. Both zttools and a cat /proc/zaptel/<device> shows no RED alarm. Is there a workaround for this? Gelson

Dear all. I recently subscribed to a VoIP provider through IAX. The require to connect with either G729 or GSM, I chose G729 based on their recommendation. The service works very well, however ... people mentions how distorted our voice sounds. We have plenty of bandwidth available so I don't think it comes from our side. What it means is that it comes from two things: 1-G729 gives bad

start it with asterisk -vvvgc bkw ----- Original Message ----- From: "Arjan" <arjan@inventionz.org> To: <asterisk-users@lists.digium.com> Sent: Sunday, July 11, 2004 12:27 PM Subject: [Asterisk-Users] Asterisk on FreeBSD 4.10 dies > Hi All, > > I'm pretty green to Asterisk. I'm trying to work towards a basic setup > with a couple of Cisco 7960's

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I'm the one who posted a message about the fact that nobody answer anymore to questions asked. I posted two days ago a problem I was facing that calls made over the Internet to my Asterisk gateway would hang-up after just 5s (no NAT were involved) Answer I got was: it's your config Well, it wasn't (as I was expecting). I

It gets definitely better every day. List of bug fixes follows: Release 1.0.5.9 7/26/2004 If SIPRegister doesn't proceed due to conditions unmet, release channel resource Fix the LED flashing issue when connection to the SIP proxy is lost. Fix the issue where the device will not resume registration when it loses connection to the outbound proxy for some time. Fixed the

Hi everybody, I have strange problem: I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone using Zap Channel) using sipgate to a number in public network. When I'm hanging up before the other person picked up the phone, the line is not closed correctly. The phone keeps on ringing until timeout (of Sipgate I assume) and it even costs my money, if the other person

Hello I currently run a few samba servers one being used as a PDC. Today I added a user to the domain and for some reason I can not get it to connect to any of the shares but "home" on the file server. % smbclient -U gregi //server3/public Password: Domain=[HYDRIX-MALVERN] OS=[Unix] Server=[Samba 3.0.28] tree connect failed: NT_STATUS_ACCESS_DENIED However I can connect with : $

Hello I used to have an extension like this which worked fine with asterisk 1.0.7 I first dial to see if an IAX phone is present, if not I would try on SIP instead exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout exten=s,2,Goto(s-${DIALSTATUS},1) ; Default action exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all. As I couldn't get to compile and run Asterisk 1.0RC1 on my default RedHat 9 I thought it was about time to upgrade to Fedora Core 2. Well, it was too late to realize the kernel 2.6 wasn't supported by Asterisk *officially* anyway. Here is what I did to get asterisk and zaptel to work under Fedora Core 2: I posted it on the wiki

I'm looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050515/d66e6335/attachment.htm

Hi there. I have a FreeBSD server running Samba 3.3, connected to a domain who's PDC is a MacOS 10.6 server running Samba 3.0.28 (ancient I know). Working all fine, except for one thing I find annoying. MacOS server has a concept of username alias. You can have as many aliases as you want, using any of those aliases are the same as using the primary one. It's rather well implemented in

Hello I am running the following setup in order to make VoIP calls at home. Home Phone <-> SPA3000 <-> Asterisk Home <- IAX2 over Internet <-> Asterisk Office The voice mail for Home Phone is hosted on the Asterisk Office machine. I wanted to have a way to check the status of my voicemail on my home phone directly. In order to do so, I've patched both Asterisk Home and

Hello I have upgraded our asterisk box from zaptel to dhadi two weeks ago... Since, there has been quite a significant amount of echo when making a call. Only for the local outgoing call, the person on the other side doesn't hear any echo. This is with a TE-110P ISDN PRI card .. I've pretty much took the original zaptel configuration and used it as-is with the dahdi one ; to no