similar to: No RED/GREEN alerts on TDM400P?

I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten => _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it?s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on

Is anyboby using the areski CDR reporting tool? I have installed asterisk-stats v1.2 three days ago, but I found a possible bug in it. My calls compare graphic shows most of the on the calls at first hours past midnight, and it never logs anything after lunch time. This is wrong, my calls are made on business hours. The call log lists those calls at the right time. Is there

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

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 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

-----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,

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'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

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 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

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

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 : $

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

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

An interesting article for those needing ammunition to sell Asterisk within their organisation or to others: "Is open source IP telephony ready for prime time? Yes" by Zenas Hutcheson, St. Paul Venture Capital Network World, 06/07/04 http://www.nwfusion.com/columnists/2004/0607faceoffyes.html On a related note, they also have an article arguing the contrary position (see link within

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the