Re: [asterisk-users] Channels freeze on Confbridge

2020-08-25 Thread Andrew Yager
On Sun, 23 Aug 2020 at 18:23, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote: > > > I had a similiar problem, but with calls dropping after 30 sec. > > It turned out that Android didn't support RP-CID (Reverse Party

Re: [asterisk-users] Remove ANSI colour trings from log files only

2020-07-24 Thread Andrew Yager
On Fri, 24 Jul 2020 at 14:23, Tim Požár wrote: > You can post process the logs with something like sed. See: > > > https://superuser.com/questions/380772/removing-ansi-color-codes-from-text-stream Yeah; we're injesting using filebeat and you can do sed style parsing on lines into it using

[asterisk-users] Remove ANSI colour trings from log files only

2020-07-23 Thread Andrew Yager
Hi, Is there a way to drop the ANSI colour strings from log files? In particular, I've got JSON logging throwing logs over to ES, but they have the ANSI colour escape sequences. Ideally I don't want to lose coloured logs from the console though, and I can't "see" a way to do this. Ast 16 at the

Re: [asterisk-users] Failed to authenticate device message

2020-07-22 Thread Andrew Yager
pdate options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- -- *Andrew Yager, CEO* *(BCompSc, JNCIS-SP, MACS (Snr) CP)* *business nbn™ advisor (advisor 01783150)* Real World Technology Solutions - IT People you can trust Voice | Data | IT Procurement | Managed

[asterisk-users] PJSIP AoR vs Endpoint

2020-07-18 Thread Andrew Yager
, because it makes X or Y amazing”. Andrew -- -- *Andrew Yager, CEO* *(BCompSc, JNCIS-SP, MACS (Snr) CP)* *business nbn™ advisor (advisor 01783150)* Real World Technology Solutions - IT People you can trust Voice | Data | IT Procurement | Managed IT rwts.com.au | 1300 798 718 *Real World is a DellEMC

[asterisk-users] 16.11.1 release removed from current

2020-07-10 Thread Andrew Yager
Hi, This is not a big issue… I just noticed a build script that was pulling the 16.11.1 release from https://downloads.asterisk.org/pub/telephony/asterisk has started to fail and went to investigate (I need to test a patch for a bugifx). It looks like when the 16.12-rc1 RC was released the

Re: [asterisk-users] AGI return codes

2013-03-20 Thread Andrew Yager
On 20/03/2013, at 10:15 PM, Ishfaq Malik i...@pack-net.co.uk wrote: I wrote the script! I've not put exit status values in it either. I've been having a look the source for res_agi_c (as a non c coder) and there is a variable called returnstatus which is instantiated with the value

Re: [asterisk-users] AGI return codes

2013-03-20 Thread Andrew Yager
Hi Ishfaq, On 20/03/2013, at 10:46 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On Wed, 2013-03-20 at 22:32 +1100, Andrew Yager wrote: Hi Andrew Thanks for the advice, I will look into it (I'm using php) The script executes successfully over 99% of the time, it is run very very

[asterisk-users] Serviced Office operator panel

2013-03-11 Thread Andrew Yager
the job, or you've seen something that works, contact me off list ASAP! Thanks, Andrew -- Andrew Yager, Managing Director (MACS Snr CP BCompSc MCP MCE JNCIA-Junos) Real World Technology Solutions Pty Ltd - IT people you can trust ph: 1300 798 718 or (02) 9037 0500 fax: (02) 9037 0591 mob: 0405

Re: [asterisk-users] Serviced Office operator panel

2013-03-11 Thread Andrew Yager
On 12/03/2013, at 10:15 AM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 11/3/13 11:07 pm, Andrew Yager wrote: Basically, if you know of a product, open or closed source, and would like to sell it to me and you think it does the job, or you've seen something that works, contact me off

Re: [asterisk-users] Serviced Office operator panel

2013-03-11 Thread Andrew Yager
Hi, It's a great console, should have included it in my list. Sadly doesn't meet SO requirements. :( Andrew On 12/03/2013, at 11:00 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 03/12/2013 12:07 AM, Andrew Yager wrote: Hi, I'm trying to find (with some desperation now

Re: [Asterisk-Users] Hylafax and Asterisk

2005-04-14 Thread Andrew Yager
Another option (which I think is just as good) is to use the patches available for chan_capi and set it up to receive faxes. Just search the list for chan_capi and fax. Yours, Andrew On 15/04/2005, at 5:11 AM, Michiel van Baak wrote: On 10:39, Thu 14 Apr 05, Kib Eki wrote: Hi, I found the

Re: [Asterisk-Users] 2 x Fritz!pci card

2005-02-07 Thread Andrew Yager
I'm actually in the process of answering this question for another person :-) Basically, it doesn't work with a 2.6 kernel. Not sure why - haven't had the time to be able to find out why either. Try centos - it's a port of red hat enterprise linux which (I think) supports the nforce 2 chipset

Re: [Asterisk-Users] iaxComm version 1.0 released

2005-01-29 Thread Andrew Yager
I think you mean http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-1.0rc1.zip :-) Andrew On 30/01/2005, at 4:02 AM, Michael Van Donselaar wrote: On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata [EMAIL PROTECTED] wrote: Hi, Is MacOSX version yet to come? Thanks to Andreas Wrede for the

Re: [Asterisk-Users] Zapata in Australia

2005-01-24 Thread Andrew Yager
As a general rule, the X100P should not be used in Australia as it is set to an incorrect impedence and can't be changed. The TDM series of cards with FXO/FXS modules can be set to work in AU. ... You should also be aware that the PSTN connect cards do not have Austel approval as yet, and so

Re: [Asterisk-Users] UPS for Asterisk

2005-01-23 Thread Andrew Yager
On 24/01/2005, at 3:26 PM, Andrew Kohlsmith wrote: On January 23, 2005 10:30 pm, Nick Bachmann wrote: HDDs don't fail because they lose power. Unless the heads crash, which can happen if power fails. I know HDD manufacturers have done head unloading and such recently, but the risk is still

Re: [Asterisk-Users] Some issues with X-Lite and codecs.

2005-01-22 Thread Andrew Yager
Hi Sergey, Have you tried phoning from X-Lite to your PSTN line, or your PSTN line to X-Lite? How is the audio quality then? Does it vary depending on the codec you have used? Andrew On 23/01/2005, at 4:31 PM, Sergey Kuznetsov wrote: Hi there, I am experiencing some issue with X-Lite. When I am

Re: [Asterisk-Users] grandstream 102 flashing

2004-10-21 Thread Andrew Yager
On 22/10/2004, at 9:06 AM, dean collins wrote: Thanks for taking the time to reply ok. The phone is sending an arp-rarp request to 67.153.142.69 The phone thinks it is ip address 192.168.0.160, I don't know how to set up a single pc to think it is the 67.153.142.69 ip address and wouldn't it need

[Asterisk-Users] DID Terminations

2004-08-18 Thread Andrew Yager
Hi, Since DID's are a topic of conversation at the moment... (and I'm in the market..) I'm looking for a DID termination in the UK (London), USA (Los Angeles) and China (Beijing). Does anyone presently use companies providing these have suggestions on who to use? We need a quality service,

Re: [Asterisk-Users] ZyXEL 2000W

2004-07-20 Thread Andrew Yager
Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 20/07/2004, at 1:26 AM, Jason Williams wrote: On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager [EMAIL PROTECTED] wrote

Re: [Asterisk-Users] chan_capi: sending incoming calls to different contexts

2004-07-17 Thread Andrew Yager
On 17/07/2004, at 6:39 PM, Stefan Reuter wrote: hi, Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? i am sending all calls to a context called capi-in where i use GotoIf based on DNID to route them. Not sure if it works for you, but the simplest way

Re: [Asterisk-Users] MWI on Grand Stream ATA-286

2004-07-16 Thread Andrew Yager
On 16/07/2004, at 11:57 PM, Dennis Cartier wrote: Sorry if this has been asked before, but does anyone have any pointers on getting the Message Waiting Indicator working on a GS ATA-286? I have tried both settings on the 286's web page for sending a SUBSCRIBE to the SIP server. I was expecting the

Re: [Asterisk-Users] Looking for WiFi phone recommendations

2004-07-16 Thread Andrew Yager
On 17/07/2004, at 7:13 AM, James H. Thompson wrote: Looking for WLAN - WiSIP - WiFi phone recommendations and experiences. What works, what doesn't. I'm using a ZyXel Prestige 2000W. It works, and sounds OK. Call hold, transfer, forward (and anything else that makes the phone worth using as a SIP

Re: [Asterisk-Users] DTMF issue --help

2004-07-16 Thread Andrew Yager
On 17/07/2004, at 3:24 AM, Eric Wieling wrote: Tony Nichols wrote: After calling a bank, or cc processing center; you have to enter your social security number, or the cc number - followed by the # key. The lovely * voice responds transfering I'm sorry that was an invalade selection. Sometimes

[Asterisk-Users] I already have a VAD frame?

2004-07-16 Thread Andrew Yager
be nice to work out why fix it... Anybody have any suggestions? Thanks, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au

Re: [Asterisk-Users] VoicePulse changes

2004-07-15 Thread Andrew Yager
Note The previous method for terminating IAX2 calls using Connect! will cease to be available at midnight (GMT) on August 15th, 2004. The message I got was at 1:51 AM EST. That means I was given negative 5 hours and 51 minutes to make this change. Check your clock. It's still July. Andrew

[Asterisk-Users] ZyXEL 2000W

2004-07-15 Thread Andrew Yager
if this feature just simply isn't implemented, or if there is likely to be something wrong with my asterisk config. Thanks in advance, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15

Re: [Asterisk-Users] Kernel panic with two Fritz cards

2004-07-15 Thread Andrew Yager
On 16/07/2004, at 9:16 AM, Nick Barnes wrote: Hi, Are you sure that you compiled the module with exactly the same gcc as the kernel? Yup - I did the kernel compile, rebooted and then did the module compilation (and then asterisk and then chan_capi etc.). No other changes were made to the system.

[Asterisk-Users] Echo issues (again...)

2004-07-11 Thread Andrew Yager
OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I Extension: Dialing: no Context:

Re: [Asterisk-Users] Echo issues (again...)

2004-07-11 Thread Andrew Yager
and to try again). Perhaps * dials too soon and interrupts the echo can doing it's training? Thanks, A _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au

Re: [Asterisk-Users] Echo issues (again...)

2004-07-11 Thread Andrew Yager
processors. A _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 12/07/2004, at 12:40 PM, Rich Adamson wrote: Andrew, What type

Re: [Asterisk-Users] Parking call problem

2004-07-08 Thread Andrew Yager
message being played. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 08/07/2004, at 8:06 AM, James Jones wrote

Re: [Asterisk-Users] Intermittent SIP 404 Not Found response?

2004-07-08 Thread Andrew Yager
On 09/07/2004, at 8:00 AM, Bruce Komito wrote: I have several SIP devices (Sipuras) that are working fine with *, except for one annoying little problem. Occassionally, after being registered for some period of time, the Sipura returns a 404 Not Found to (I assume) an INVITE request. Of

Re: [Asterisk-Users] Unreliable dtmf digit generation from tdm400p

2004-07-07 Thread Andrew Yager
On 07/07/2004, at 7:20 AM, Alok K. Dhir wrote: I have the *exact* same problem. Please let me know if you have found any solution. Thanks! In my setup I have 2 of the TDM400P cards, with four FXO modules each. I'm not having this problem on either of my TDM400 cards with a mix of FXO and FXS

Re: [Asterisk-Users] How to differentiate incoming calls with grandstream phone

2004-07-06 Thread Andrew Yager
in the sip.conf file for each of the phones. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ smime.p7s Description: S/MIME

Re: [Asterisk-Users] Calling an outside phone number as part of a hunt

2004-07-05 Thread Andrew Yager
No warranties on this working - wrote it off the top of my head. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 06/07/2004, at 12:51 PM

Re: [Asterisk-Users] SPA-2000, call for help testing echo issues...

2004-07-04 Thread Andrew Yager
On 04/07/2004, at 6:32 PM, Trevor Peirce wrote: Mike Benoit wrote: I'm curious to know if anyone else using SPA-2000's have the same issues. I wonder if when calls are made from SPA-2000's to PSTN numbers through Asterisk, asterisk is just amplifying the SPA-2000's own echo somehow. I've

Re: [Asterisk-Users] What is IAX Trunking?

2004-07-04 Thread Andrew Yager
On 04/07/2004, at 11:24 PM, Deon Rodden wrote: Sorry, I've been on voip-info.org but I still can't get a clear definition of what IAX trunking is. It says you need the timing from a zaptel device (or ztdummy or zaprtc) to make it work, but nothing specific about what it is or what it does.

Re: [Asterisk-Users] Cisco 7960 Reboots when SoftPhone calls it?

2004-07-04 Thread Andrew Yager
On 04/07/2004, at 11:26 PM, Deon Rodden wrote: I have a Cisco 7960 phone, I just updated to P0S3-07-1-00 image/firmware, due to a ton of fixes from P0S3-06-3-00 which we were running. But now when I call my phone using X-Lite, the second I answer, it reboots. I tried upgrading to the latest

Re: [Asterisk-Users] What is IAX Trunking?

2004-07-04 Thread Andrew Yager
Is this done automatically when using IAX2? You need to specify trunk=yes in the IAX config file. Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Grandstream 1.0.5.30 available

2004-07-02 Thread Andrew Yager
On Fri, 02 Jul 2004 17:33:30 +1000 Master Abi [EMAIL PROTECTED] wrote: New firmware version at http://www.hellofone.com/downloads.html. Might fix the no register issue and others. I tried this a few days back, totally hosed my phone, has to back to Grandstream, don't touch it! -Max. I just

[Asterisk-Users] Zaptel, Line Impedence and Echo

2004-07-02 Thread Andrew Yager
, and I know that _opermode should be set to 3 and opermode should be set to AUSTRALIA. Can someone with more knowledge enlighten me as to how I might go about doing this - or indeed if I need to? (If it helps, I'm running Fedora Core 1) Thanks in advance, Andrew _ Andrew

Re: [Asterisk-Users] Zaptel, Line Impedence and Echo

2004-07-02 Thread Andrew Yager
Is there a way to specify that info for the zaptel init.d script? (as a side note - I'm talking on my Cisco 7960 via the FXS now, and it sounds fine) Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945

Re: [Asterisk-Users] Zaptel, Line Impedence and Echo

2004-07-02 Thread Andrew Yager
On 02/07/2004, at 10:43 PM, Andrew Yager wrote: Is there a way to specify that info for the zaptel init.d script? In answer to my own question - yes there is. You should modify /etc/init.d/zaptel and find the line that reads insmod ${x} ${ARGS} Change it to read: insmod ${X} opermode=AUSTRALIA

Re: [Asterisk-Users] Zaptel, Line Impedence and Echo

2004-07-02 Thread Andrew Yager
On 03/07/2004, at 7:15 AM, Richard Scobie wrote: [EMAIL PROTECTED] wrote: Does this only work with the new fxo modules? Yes. I don't have one of the new new modules (I have a Rev C) - and I seem to notice a difference - am I just seeing things? What is a new FXO module these days? Andrew

Re: [Asterisk-Users] Modems behind Asterisk - how?

2004-06-29 Thread Andrew Yager
Hi, Looks like your message got lost in the thread. On 29/06/2004, at 8:47 AM, John Vogel wrote: 1. Use 4 Sipuras (approx. $400). Only problem is, I can't get this to work! Sipura says use the G711 codec but it's not working for me. Anybody have this working? I haven't got any Sipura's to test,

[Asterisk-Users] Complaining Emails

2004-06-29 Thread Andrew Yager
as much as they do about this software, and be able to provide the same level of help to other people. Let's not spoil what we have. Just my 2 cents... Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566

[Asterisk-Users] Latest CVS fax detection grandstream bug

2004-06-25 Thread Andrew Yager
going to play with the different options (incoming/outgoing/both) to see if it makes a difference. Thanks, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au

Re: [Asterisk-Users] Latest CVS fax detection grandstream bug

2004-06-25 Thread Andrew Yager
to do from here would be useful. Thanks, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 25/06/2004, at 7:16 PM, Andrew Yager

Re: R: [Asterisk-Users] Which Linux ?

2004-06-24 Thread Andrew Yager
On 24/06/2004, at 4:48 PM, Manuel Wenger wrote: Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. Just my experience: I have installed Asterisk twice on Fedora Core 1 with kernel

Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!

2004-06-24 Thread Andrew Yager
Just to further update this issue: Three simple source code changes (testing-only at this point) resulted in zero detectable echo on all incoming and outgoing tdm pstn calls, even in the first second or two. I've not noticed any unusual side effects at all. Since the changes were only to

Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!

2004-06-24 Thread Andrew Yager
Well there you go... No I hadn't done that. And it still sounded OK. Which is quite bizzare. Yesterday I has having terrible echo issues. Today none at all. *sigh* I'll see what setting echotraining=800 does for me... Andrew _ Andrew Yager Real World Technology Solutions

Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!

2004-06-24 Thread Andrew Yager
_ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 25/06/2004, at 1:35 AM, Ryan Courtnage wrote: On Thursday 24 June 2004 09:01, Rich Adamson wrote: Per

Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!

2004-06-24 Thread Andrew Yager
On 25/06/2004, at 12:57 AM, Andrew Yager wrote: I'll see what setting echotraining=800 does for me... It still sounds good. There was no noticeable echo on the three calls we tried. Difficult to say whether it is a greater improvement or not, but I'm sure I'll have a feel for it after tomorrow

Re: [Asterisk-Users] chan_capi problem - hangup???

2004-06-24 Thread Andrew Yager
you the context the controller puts you into does not have an exten s= handler, nor does it have an invalid handler (exten i= ). You can discover this context by examining the context= line in your capi.conf Create one, and see if that fixes the problem. Andrew _ Andrew

[Asterisk-Users] Latest CVS, Grandstream and Zaptel bug?

2004-06-24 Thread Andrew Yager
this at yes, but when changed it to no to test busycount=4 musiconhold=default faxdetect=incoming channel = 1 Any help or suggestions on what to try or where to go would be appreciated. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph

Re: [Asterisk-Users] Latest CVS, Grandstream and Zaptel bug?

2004-06-24 Thread Andrew Yager
Already set. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 25/06/2004, at 3:38 PM, Peter Boot wrote: I had the same problem

Re: [Asterisk-Users] Busy message

2004-06-22 Thread Andrew Yager
the relevant section of your extensions.conf . Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 22/06/2004, at 2:26 PM, Simon Brown wrote

Re: [Asterisk-Users] Need different contexts for 2 X100P FXO Cards and forwarding calls

2004-06-21 Thread Andrew Yager
_ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 21/06/2004, at 11:40 PM, [EMAIL PROTECTED] wrote: Thanks Andrew, I will give it a try. I assume I can put channel

[Asterisk-Users] Outgoing call via Fritz!

2004-06-08 Thread Andrew Yager
likely to originate from? Us or our telco (which is Telstra)? Thanks, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au

[Asterisk-Users] FXO answering quicker

2004-06-06 Thread Andrew Yager
Hi, I don't know if this is possible - but can I set up asterisk to answer the FSO line after one or two rings rather than four? I haven't (yet) found a configuration variable to let me do this... Thanks in advance, Andrew _ Andrew Yager Real World Technology Solutions

Re: [Asterisk-Users] FXO answering quicker

2004-06-06 Thread Andrew Yager
a single ring and then answer. Is that possible to do? Thanks, Andrew On 06/06/2004, at 5:37 PM, Nicholas Bachmann wrote: Andrew Yager wrote: Hi, I don't know if this is possible - but can I set up asterisk to answer the FSO line after one or two rings rather than four? I haven't (yet) found

[Asterisk-Users] Compiling on OSX 10.3.3

2004-05-25 Thread Andrew Yager
machine is: Darwin hostname.goes.here 7.3.0 Darwin Kernel Version 7.3.0: Fri Mar 5 14:22:55 PST 2004; root:xnu/xnu-517.3.15.obj~4/RELEASE_PPC Power Macintosh powerpc Yours, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945

[Asterisk-Users] NetJet and RAS

2004-05-23 Thread Andrew Yager
a silly PBX...) Thanks in advance, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) http://www.rwts.com.au/ _ smime.p7s Description: S/MIME cryptographic signature

Re: [Asterisk-Users] NetJet and RAS

2004-05-23 Thread Andrew Yager
FXS and one FXO. Others in Australia are also using Asterisk with the Zaptel cards. Regards, Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Yager Sent: Monday, 24 May 2004 9:37 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NetJet

Re: [Asterisk-Users] NetJet and RAS

2004-05-23 Thread Andrew Yager
/wiki-Asterisk+Hardware in particular covers hardware such as cards. Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Yager Sent: Monday, 24 May 2004 9:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NetJet and RAS Thanks! That's good