similar to: SIP Provider Question

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

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 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

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

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

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

Index: channels/chan_alsa.c =================================================================== 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/

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

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

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

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

Residential Long Distance. One of my biggest pushes towards a VoIP provider was cheap long distance. Now in the U.S. at least with SBC they now have a plan for Unlimited Long Distance. The price is 30.00 a month if you do not have a couple of required features on the line already like Caller ID and a feature like 3 Way or Call Waiting etc. in which case that lowers the price to 20.00 a month.

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want