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Full_Name: Chris Paulse Version: 2.1.1 OS: WinXP Submission from: (NULL) (129.98.60.134) This seems strange. I have a small block of code that repeatedly calls rbinom. I put a break in there in case it returns NaN, as I've been having problems with this. Here is a transcript from the debug session: Browse[1]> theP [1] 1 Browse[1]> yleft[dataIndex] [1] 3 Browse[1]> rbinom(1,3,1)

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-w...

...WARNING[3491] app_dial.c: Had to drop call because I couldn't make Zap/1-1 compatible with SIP/usa-e2ea It only happens with sip trunks, with iax2 trunks the calls works like a charm. Can you help to fix it? -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : 103@sip.manta.telconet.net e-mail: gsalas@manta.telconet.net www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://w...

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

...148|FUNC |GLOB |0 |9 |_catopen [6100] | 525792| 16|FUNC |WEAK |0 |9 |_cleanup [7789] | 651320| 24|FUNC |GLOB |0 |9 |_creat64 [8799] | 515852| 56|FUNC |GLOB |0 |9 |_ctermid [8814] | 271940| 72|FUNC |GLOB |0 |9 |_ctime_r [8071] | 515964| 144|FUNC |GLOB |0 |9 |_cuserid % /CP Lin

...1: XBUS-00/XPD-00 "Xorcom XPD #0/0: FXO" ;;; line="1 XPP_FXO/0/0/0 FXSKS" signalling=fxs_ks callerid=2627839 group=1 context=from-zaptel channel => 1 Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas at manta.telconet.net www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : 666 at sip.manta.telconet.net FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or...

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on

...9 (1785868). Clear? yes Entry '00013' in /2/8176 (2998284) is a link to directory /1/8045/00036 (1261580). Clear? yes Entry '00013' in /2/8178 (3047436) is a link to directory /1/8039/00036 (1196044). Clear? yes Entry '00009' in /2/8184 (3227660) is a link to directory /1/8071 (1949708). Clear? yes Entry '00012' in /2/8182 (3293196) is a link to directory /1/8029/00037 (1277964). Clear? yes Entry '00012' in /2/8193 (3375116) is a link to directory /1/8021/00037 (1228812). Clear? yes Entry '00008' in /2/8188 (3522572) is a link to directory /1/8...

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Hello group, I'm running asterisk @ home 1.5 - I would like to change these messages(call attend) to Spanish, how I can do that. Thanks, Nelson

...hone, in the product page http://isupport.lge.co.kr/html/ibu_lgic_modelView.jsp?jgrcode=D2_IPTP&modelid=M_IP100C the features are showing SIP and H.323 support. Can be used with my asterisk box? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : 105@sip.manta.telconet.net e-mail: gsalas@manta.telconet.net www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://w...

Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... Thanks bye

Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Many thanks, Nathan -- -------------------- www.nathanpralle.com --------------------

...2.14/codecs' make: *** [subdirs] Error 1 ruidoso:/usr/src/asterisk-1.2.14# uname -sr Linux 2.6.8-12-amd64-k8 Maybe I'm forgotting to install any dependency? Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-w...

I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david@safedatausa.com

...evice /dev/dsp: No such file or directory -- Executing [2@default:1] BackGround("OSS/dsp", "demo-moreinfo") in new stack << Console call has been answered >> -- <OSS/dsp> Playing 'demo-moreinfo' (language 'en') [Apr 24 12:34:36] WARNING[8071]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Any idea why I can't hear the asterisk default demo when dialing 2? - sf

I am looking for a retail DID provider which should provide unlimited incoming calls something around 4-5 bucks . Nufone seemed like a good choice at $5 but they are down again :( . Any suggestions please . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070802/087c4117/attachment.htm

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: