similar to: G729 or GSM

-----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,

-----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

-----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

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

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 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, 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

Hello Steve. On 17/05/2005, at 10:54 PM, Steve Underwood wrote: > When a fax is received the header you see is part of the image. As > such, it ends up in the TIFF file as part of the image. It is not > available as text anywhere. The only way to make it available as > text would be to implement OCR. That is complex and messy, and I > have no intention to try. The only

Hello I upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10) And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one. In the UPGRADING.txt in Asterisk it says: * The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key)

I saw somewhere that the last kernel to work properly with the zaptel drivers when using data over it was 2.4.20. Has this been since fixed to work with newer kernels? -Mike

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

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

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

Dear all. Okay, I'm about to give up now as this issue has been driving me insane for a while I have an asterisk server (which I have just updated to 1.2beta1). Several phones are connected on LAN, no problem People on the go using either x-lite or eyebeam without a problem I have a Sipura 3000 configured at my parents in France used to work fine until about 1 month ago. Since it can

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

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

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

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