search for: timebandit001

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

...run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Sat, 1 Sep 2007 12:14:49 -0400 > From: "Time Bandit" <timebandit001 at gmail.com> > Subject: Re: [asterisk-users] Cisco Directory Format > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <1e2050d50709010914h763bce70xb39c5fdd20f53b65 at mail.gmai...

Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the "Qs about FXO/FXS cards" thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi

Hi List, I am a newbie, just came to know about asterisk a few days back while installing suse 9.2. I have a question for which I am sorry to say,but I havntread through all the archives, but AFAIK i didnt get the answer in the archives. My question is like this. I am in Nebraska, US. I have broadband cable connection at my home. And I have friend and family in other country. Using asterisk

...I'm realy happy to hear that, and as a matter of fact, all my Asterisk hardware is Intel server products, from chasis to MB. I know I can trust this hardware and I have excellent support from the guys I buy from. Thank you. >Message: 5 >Date: Mon, 3 Jan 2005 15:48:47 -0500 >From: <timebandit001@gmail.com> >Subject: Re: [Asterisk-Users] 8 pstn lines+ on Asterisk supported > hardware. >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <1e2050d50501031248651f17c6@mail.gmail.com> >Content-Type: text/pl...

Hello All, I am going to try and install Asterisk on a server machine that I have and wanted to get the opinion of experienced people as to the best method. Initially I was going to purchase the Asterisk Starter Kit at: http://www.thevoipconnection.com/store/catalog/product_16215_Asterisk_Starter_Kit.html But from what I can find out in the reviews, the book that comes with the kit is not

Hello list, I am having trouble listening to GSM files created by Asterisk using a browser. I am noticing that some of my users succeed in listening to them and some others don't. I guess it is something of a codec problem that does not seem to be installed on all machines (though they are all WinXP). Anybody knows what one should do to listen to GSM files? I send files through the

Nuthin beats an Atlas: http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8 A.pdf Telephony Swiss army knife. You can make it do anything. Be prepared to crap your pants when you see the price, though. -----Original Message----- From: Time Bandit [mailto:timebandit001@gmail.com] Sent: Friday, May 26, 2006 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] End of migration: adding support for some analog phones > I'd like to keep some of analog lines to support modem, fax and some older > stuff. What'...

From: "Time Bandit" <timebandit001@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date: Tue, 7 Nov 2006 08:53:51 -0500 Subject: Re: [asterisk-users] Follow Me problems > Today we appear to have discovered our first bug. We have an extension > se...

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29 PM Subject: Re: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX >I did change the port 4569. > Also my router...

...be changed in the apache conf file if wished. > > Let us know how you go. > > C > > -----Original Message----- > From: World Wide Web Owner [mailto:www@mail.one2net.co.ug] On Behalf Of Julius Kidubuka > Sent: 03 March 2005 09:45 > To: asterisk_list@burntwires.com > Cc: timebandit001@gmail.com; asterisk-users@lists.digium.com > Subject: Getting phpconfig to work? > > Hi, > > I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; > > Index of /phpconfig > Na...

...be changed in the apache conf file if wished. > > Let us know how you go. > > C > > -----Original Message----- > From: World Wide Web Owner [mailto:www@mail.one2net.co.ug] On Behalf Of Julius Kidubuka > Sent: 03 March 2005 09:45 > To: asterisk_list@burntwires.com > Cc: timebandit001@gmail.com; asterisk-users@lists.digium.com > Subject: Getting phpconfig to work? > > Hi, > > I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; > > Index of /phpconfig > Na...

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the "great white north" (sources, services, telcos, etc.), I created the asterisk-canada mailing list: http://lists.syonex.com/mailman/listinfo/asterisk-canada or asterisk-canada-subscribe@lists.syonex.com Cheers! John

Hi My card is working, but when I reboot the machine, most of the times it is not working, I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)" To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose "Ignore" and then it's

...;> >> Let us know how you go. >> >> C >> >> -----Original Message----- >> From: World Wide Web Owner [mailto:www@mail.one2net.co.ug] On Behalf Of >> Julius Kidubuka >> Sent: 03 March 2005 09:45 >> To: asterisk_list@burntwires.com >> Cc: timebandit001@gmail.com; asterisk-users@lists.digium.com >> Subject: Getting phpconfig to work? >> >> Hi, >> >> I have just tried to get phpconfig to work but to no avail. In my >> browser >> I type; http://ip-of-machine/phpconfig/ and this returns the following >>...

Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP function and assign it to a variable in an AGI script. I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's treating DUNDILOOKUP literally, rather than calling a funciton. I've tried this: EXEC "Set" "DIALPATH=${DUNDILOOKUP(2944093|180net)}" and also: SET VARIABLE

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

I have been looking around for Outlook Integration for Asterisk. Saw the Asterisk TAPI wiki page and also ran across this: http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) It looks like Fonality has managed to make an app that does screen pops and allows click to dial. Has anyone else been able to get this all to work successfully? Looks pretty slick.