search for: mcnamara

...rcle.svg) before the text. I've found that I can plot a .png image inside the plot boundaries using rasterImage() but I can't figure out how to do it outside the boundaries. Any help would be great. If you know unicode or Adobe Symbol encoding for the CC logo, that might work too. ~Amelia McNamara Statistics PhD student, UCLA

hi I recently spoke to mr McNamara on IRC, and he mentioned there was a "far better way to do realtime-stuff than the usual realtime in asterisk, and that this was GPL". He failed, however, to ever mention how this could be done, so I just wonder if someone else might know... ? roy -- Roy Sigurd Karlsbakk roy@karls...

I looked on google for some time and haven't found a source for 9-911 labels to attach to phones. I have run into them in the past but I can't currently find a source. I am looking for a low volume were having them printed up for me would not be economical. Any suggestions would be welcome. Thanks -Jason -- ----------------------------------------------------------------------------

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

...distribute it under certain conditions. See the GNU General Public Licence for details. As far as I can see all files are updated, so nothing is lost, but it is wasting bandwidth which is one of the great things about rsync! Anyone seen this before? Anyone know if there are fixes for it? Frank McNamara MCH Konsulting Sweden frank@abacus4.com _________________________________________________________________ Chat with friends online, try MSN Messenger: http://messenger.msn.com

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi list, I'm interested in receiving incoming call to my Asterisk PBX thru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitation such as much few states that a call can originate? How do

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

Hi, My question is in refernece to the posting by Jeremy McNamara here.. http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html He states that in order for "trunking" to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my &qu...

Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? Gene Kochanowsky Solution

Hi all Asterisk Gurus. I am really badly in need of help. Asterisk is very lovely software, but has one big disadvantage.. lack of documentation.But let's get to the point. 1. Is it normal that I get such a crappy quality with iax, some drops and clicks? Could anyone with some similar setup check my quality and say if this is what the people are so excited about? ( I used to work as a speech

Hello all, Please forgive me if this sounds a little (or a lot) ignorant as I am very new to asterisk. Right now I have two pc's connected back to back through an E100 card running asterisk. I have openh323 running as well and I am able to route calls through the E1 line. Next up I would like to be able to register asterisk with a gatekeeper. On another computer is running openGK. Using

I'm having great success using Cisco 7960s with Asterisk but one Cisco 7960 specific function that's puzzling me is programming the 'line buttons' (6 buttons on the upper right side of the phone) for speed dial. I'm not using any Cisco applications like Call Manager and the phones are SIP. *All* of the documentation I've found over the last several days continues to refer

...dispute the charges today. > It is very obvious you edited that screen shot. Processing fees are not refundable as defined in our terms and conditions, which you agreed to prior to signing up. $10 is available for refund, that is all and we will own rights to your toll-free number. Jeremy McNamara > Jeremy McNamara wrote: > >> Linn Boyd wrote: >> >>> Close our account fractionit and refund all money that we have paid >>> you, you did not disclose that it would take 7-10 days before it was >>> provisioned before we ordered the number, had that...

Hello everybody, First time installer and I need the lists advice. My plan is to use asterisk PBX with some hardware to terminate my calls coming from several operational gnugk gatekeepers. Do have RH9 and downloaded the latest asterisk from CVS. Compiled according instructions and is running fine. Could hardly find any info on h323 implementation untill the REAME in the channels directory.

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

Hi, I have recently discovered the project along with the PhoneJacks produced by quicknet, they could be the answer to something I have been looking into. I would like to be able to test using a dial-in server & possibily also a Windows RAS server, however I only have 1 phone line. I was thinking that I create a setup like that illustrated below to solve the problem:- Ext 1000

Hi, Has anybody managed to get callerid properly set on a call from local to asterisk SIP endpoint through h323-pstn gateway to a regular phone. I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it. When I place a call to pstn I'm not receiving 12125551234 as the clid, but a number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other

Hello all ! How can I make conference authorization based on pin number ? I have: exten => 1,1,Meetme,1234|ps|2222 where 2222 is a pin number and this doesn't works Where do I have to add information about pin number ?? Greetings Andrzej Radke

All - NEED HELP BADLY.. Using Asterisk with H323 Channels.. Everything is installed but we cannot register w/ Lucent gatekeeper. We ran ethereal and found that it was making GRQ (Gatekeeper discovery requests).. We had provided the name of the Gatekeeper (it's IP) and cannot determine why it's trying to do a GRQ. We want it to go straight into RRQ or ARQ and skip the GRQ.. Netmeeting does