search for: bielicki

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

...9; isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Thanks, Chad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Bielicki Sent: Saturday, September 18, 2004 9:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system try zaprtc from www.junghanns.net. Works fine in my SMP systems ----- Original Message ----- From: Chad Brown <chad.brown@identitymi...

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...

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

...orzaprtc It was my understanding that you don't 'disable' rtc, but recompile it as a kernel module. Again, just my understanding as I can't try it until monday. Matthew ----- Original Message ----- From: "Chad Brown" <chad.brown@identitymine.com> To: "Michael Bielicki" <cypromis@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, September 19, 2004 1:13 PM Subject: RE: [Asterisk-Users] Timing source on SMP system - Disable RTC forzaprtc Does anyone know where to disabl...

I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the

I am using a grandstream phone with g729 and alaw odecs and in both modes I cannot seem to pass dtmf's, neither inband nor out of band, neither wthrough a lcoal server nor through a natted connection. Am I missing something ?

Hello, One quick question. Does anyone has experience implementing unified messaging (UM) using Asterisk. Does Asterisk has support for UM ? Thanks, Tarun ___________________________________________________ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2

This is BAAAAAAAD! Now even SIP get's "tainted"... http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95 _________________________________________________________________ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream

...sers-admin@lists.digium.com > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Asterisk-Users digest..." > > >Today's Topics: > > 1. Re: CE certification for Europe (d hinton) > 2. Re: MP3player problem (Michael Bielicki) > 3. Re: FAX over IAX (Steve Underwood) > 4. Re: CE certification for Europe (Mark Spencer) > 5. Re: CE certification for Europe (Steve Underwood) > 6. Re: MP3player problem (Martin Pycko) > 7. Re: MP3player problem (Martin Pycko) > 8. Re: FAX over IAX (Martin Pycko)...

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

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

...nnels/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/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality 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.

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...

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.

...an offer for 30 black BT101's delivered to Warsaw/Poland for end of next week. Additionally 3-5 Cisco 7960's depending on price I need price including shipping and delivery guarantee. Delivery would be to: Halo Kwadrat Sp z o.o. Ul. Konwiktorska 7/9 Warszawa 00-216 Poland cheers Michael Bielicki PS: a asterisk-biz list would not be the worst of ideas

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:

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif