search for: taan

Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly

Is it possible to give a false ringbakc on asterisk ? -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or...

Hi folks: Does the TDM400 card from Digium only support FXS, or is FXO functionality available or planned? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming, Networking

...========================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_alsa.c,v retrieving revision 1.2 diff -r1.2 chan_alsa.c 1042c1042 < if ((cfg = ast_load(config)) { --- > if ((cfg = ast_load(config))) { -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or...

Anyone tested this card in Greece? I would like to buy several of these but I don't know if it works well with the PSTN in Greece. The Quicknet Linejack card seems to be unable to understand the telephone network's tones (busy, ringing,...). TIA, Michael.

Hello. Well I now have asterisk installed. I've printed out the asterisk web site. I've printed the draft Asterisk handbook V2 I've printed the Introduction to the asterisk open source pbx Because I'm experimenting, I would like to do things in a certain order :- 1) VOIP inside the private LAN from one computer to another. e.g. 192.168.1.1 to 192.168.1.2 etc. 2) VOIP

is there any particular reason why there is no option to configure the codec framesizes in iax2 ? It would come rathrer handy to decide if you want less bandwidth or more robustness on the payload side ... -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or...

Hello all, A newer version asterisk-oh323 (v0.5.5) is available. This version contains updates for compilation with latest Asterisk sources, some additions ("atexit" cleanup) and some other fixes. http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

Hi, I am trying to set up * and ISDN-PRI (channels 1 - 15) using E100 boards. I installed zaptel and libpri. When I execute modprobe -r wct1xxp I get an error message: ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows my /etc/zaptel.conf:

If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. It's a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not

Mythical Asterisk Creatures, oft-discussed, rarely seen: 1) An "advanced" graphical user interface 2) An IAX2 hardware device 3) A Radius CDR report module 4) A live-method, robust SQL-based dialplan 5) LDAP/SQL/Radius authentication for SIP phones 6) Robust R2 signalling support 7) Multilingual language recordings of all existing * .gsm files 8) Free exchange of PSTN gateways in

Hi, Below if the error message which I got from asterisk. I was trying to fax to asterisk from my fax machine. I really dunno what is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone help me what could be the problem. -- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack -- Goto (13732,s,1) -- Executing

1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet->PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even

We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly "standard" EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back

Hi, I am starting to look at the costs involved in putting a PBX system together using *.. We will be using ISDN lines in the UK, which when installed provide a box on the wall with 2 analog ports and 2 digital ports, so these seem to be the options available to me.. 1. I could use an ISDN board but I have not really read good things on the mailing list about using ISDN boards, and I suspect

Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got "drafted" to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming

...___ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > >--__--__-- > >Message: 2 >From: Michael Bielicki <Michael.Bielicki@global-gateway.net> >Organization: TAAN Consultants Ltd >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] MP3player problem >Date: Thu, 3 Apr 2003 17:18:55 +0100 >Reply-To: asterisk-users@lists.digium.com > >what does playback use ? > >On Thursday 03 Apr 2003 16:18, Martin Pycko shaped the electr...

Hi, I have a genotype data for both case and controls and would like to calculate the HW p-value. However, since the number of one genotype is 0, I got wired result. Would someone help me to figure it out? Or confirm it's right? Thanks a lot. ============ > library( "genetics" ) NOTE: THIS PACKAGE IS NOW OBSOLETE. The R-Genetics project has developed an set of enhanced

Many of you are familiar with how lousy Festival sounds. AT&T has a product, NaturalVoices, that sounds much better. There are male & female voice fonts for US/UK/Indian English, French, Spanish, and German. I am considering offering a linux-based text-to-speech engine based on the NaturalVoices runtime. An asterisk module would also be provided, making it easy to add natural sounding