search for: jbryan

...: yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips : 2924.54 == Thanks for any help or advice anyone may have. Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. jbryan at productiveconcepts.com (765) 964-6009, ext. 224

...ce for your help, sorry for all my replies, but i`m not a regular user of the list, and i don't lnow how to make replies yet :( > > Regards > > > ------------------------------ > > Message: 5 > Date: Sun, 7 Feb 2010 20:52:14 +0000 > From: "Josiah Bryan" <jbryan at productiveconcepts.com> > Subject: Re: [asterisk-users] Asterisk going down > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > > <2126791352-1265575971-cardhu_decombobulator_blackb...

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for

2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

...te the %N for the number clicked. Assuming you've chmod'ed +x the script and set the server, user, and secret correctly, when you click on a contact's number, your phone will ring. Any questions, let me know! Cheers! -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. jbryan@productiveconcepts.com (765) 964-6009, ext. 224 -------------- next part -------------- A non-text attachment was scrubbed... Name: asteriskdial Type: application/x-perl Size: 2148 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050509/80a4e164/asteris...

...of inodes on this disk. We have been running EXT3 fs for sometime now, and recently upgraded kernels from 2.4.10-ac11 to 2.4.16 sda1 is a RAID5 hardware array (DPT PM1554U2). Can anyone make out what these errors mean? -- ----------------------------------------------------- Jessie Bryan e - jbryan@netlojix.com w - www.netlojix.com -----------------------------------------------------

Hi all, I know when doing a Dial, when the called party hangs up, we have a few different ways to redirect the calling party to other parts of the dialplan. In this case, I have someone who would like to do the opposite... When the calling party hangs up after a Dial(), redirect the called party to another location. I'm not sure how else to describe what the user wants to do, but I'm

Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

hi, I recently changed my filesystem to ext3 (from ext2), and since then, my hard disk sometimes seems to 'hang' a few seconds (in fact the whole system hangs then, and only the hd LED is on), before continuing normally... This happens for example (not always though) when I untar an archive and try to ls in the newly created directory... Is there an explanation for this? And if so,

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Hello, I am a newbie in Asterisk IP PBX but I am very impressed by its functionalities. I have read that It can work over IP network and across the PSTN. I am not very sure how it works over the PSTN.. In case if people have not yet the Internet or SDL access, I would like just to know if it is possible to interconnect two IP LANs using the traditionnal analog Network and Asterisk PBX? In such I

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

Wanted to make openssh w/both -lwrap and BIND 8.2.3's -lbind on an m68k NeXT 3.3. Started w/then current openssh-2.3.0p1. Had to use -posix for compiler and loader to get -lbind to work, and had to throw out all openssh NeXT porting to get -posix to work. By the time I had it all working and installed, a security mailing list said there was a new openssh version released. Bagged