Re: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia

I had a similar issue both with the X100P clones and TDM400. Both were fixed by enabling AU zone and the busydetect functions. Don't forget a full asterisk reload needs to take place after changing Zap conf files, not just a soft-reload. Best way is to reboot the computer. Mike I have a

Re: [Asterisk-Users] oh323 on @homeasterisk

Can you please detail the steps you have taken to successfully compile this on @home asterisk? Regards Mike - Original Message - From: CM Rahman Jr. [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 09,

Re: [Asterisk-Users] Set system time over the phone

PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 05, 2005 8:02 PM Subject: Re: [Asterisk-Users] Set system time over the phone On Tue, Apr 05, 2005 at 09:45:54AM +1000, Mike Sander wrote: I have installed Asterisk using the [EMAIL PROTECTED] image for a client that is VoIP-a-phobic

Re: [Asterisk-Users] Set system time over the phone

Looks good - thanks for the help! Mike - Original Message - From: Roman Volf [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 05, 2005 4:48 PM Subject: Re: [Asterisk-Users] Set system time over the phone

[Asterisk-Users] Set system time over the phone

I have installed Asterisk using the [EMAIL PROTECTED] image for a client that is VoIP-a-phobic. Hence the system cannot be connected to their LAN at all - don't ask why! I have tested the clock at my installation lab, and all is fine, but they might want to set/check it. I know there is the

[Asterisk-Users] Asterisk@Home H323

I am looking for a step-by-step on adding H323 to [EMAIL PROTECTED] So far I have installed [EMAIL PROTECTED], upgraded to the CVS-HEAD and followed instructions according to voip-info and this list's archives. I keep getting critical errors on compilation of H323, both Open 323 and OH323. Has

Re: [Asterisk-Users] How to do something random?

Logically, you should build something like this: 1. Pick a number between 1 and 3 2. Save the number to a variable indicating which line you are about to try 3. Check if it's free, if so make a call 4. If not, pick a number between 1 and 2 5. Make sure you haven't tried this number before (a loop

Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons?

You can share them here: http://asterconf.hopto.org/ Mike - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Nicolás Gudiño [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 12:10 AM

Re: [Asterisk-Users] OT: Best DB

But Budwieser tastes like water to most Australian beer drinkers. (Now I'm in trouble!) Mike - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 18, 2005 11:48 AM

[Asterisk-Users] New Help Site - cut down on Mailing List questions

of posts for people looking for instant setups, who don't want to use AMP or otherwise. That way, we can return this list to the discussion of Asterisk issues, rather than just a startup resource and helpdesk. I'm always interested in anyone's comments. Cheers Mike Sander sanderm at iprimus.com.au +61

Re: [Asterisk-Users] Dock-n-talk connection to asterisk

Hi Peter. Look in last weeks (1/3/05) Sydney Morning Herald Tuesday IT liftout. They talk there about GSM gateways. It was made by Ericson I think, for around $1000. It's not meant for computer, rather as a FXO/FXS gateway to plug your house phone in for exactly the purpose you are talking

[Asterisk-Users] New Help Site - cut down on Mailing List questions

issues, rather than just a startup resource and helpdesk. I'm always interested in anyone's comments. Cheers Mike Sander [EMAIL PROTECTED] +61 2 401 010 289 (Australian mobile) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls

I'm sure this has been said, but the [EMAIL PROTECTED] installation of Flash Operator Panel shows the handset shaking when a phone is ringing, so there is a way to do it. I'd search in there. Mike - Original Message - From: mattf [EMAIL PROTECTED] To: 'Asterisk Users Mailing List -

Re: [Asterisk-Users] Autodetecting faxes

That's all very well, but what do you do if you only have SIP extensions and IAX trunk - no Zaptel card. Will Fax detection still work at all? Thanks Mike - Original Message - From: Adrian Chapman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion

[Asterisk-Users] How to download CVS with attended transfers

Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the

[Asterisk-Users] How to download CVS with attended transfers

Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the

RE: [Asterisk-Users] Multi Asterisk Server Transfers

I believe this is what I have, but it still insists on running the transfer from the head office. Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head office

RE: [Asterisk-Users] Multi Asterisk Server Transfers

Simple as that? Anyone know a good IAX phone (not softphone)? Thanks Mike Then you need to use the same protocol to the provider. One office is using SIP, the other is using IAX. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database:

[Asterisk-Users] Multi Asterisk Server Transfers

Hi, We are in the business of setting up * servers for businesses, attached via IAX trunks to our VoIP provider (also using *). I have a client with a head office * server, who wants a number of remote offices, with just 1 SIP connection to each. I can arrange this no probs with our providers,

RE: [Asterisk-Users] Multi Asterisk Server Transfers

I agree with you. If every office had a * server, it would be fine. i.e. Office 1 rings office 2, then gets transferred to office 3, then connection is direct from office 1 to 3, and 2 releases all contact. However, what if office 3 is a 1 person office, with just a single SIP phone connected to

[Asterisk-Users] SetGroup and CheckGroup problems

is clear, begin dial sequence exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2}) Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 24

[Asterisk-Users] SetGroup, CheckGroup

is clear, begin dial sequence exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2}) Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 24

RE: [Asterisk-Users] Call Pickup

is clear, begin dial sequence exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2}) Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 24

RE: [Asterisk-Users] SetGroup and CheckGroup problems

external calls, it shows 2, but there doesn't seem to by rhyme or reason. It makes sense to me to show 1 when they are on the call, 2 when they have 2 going etc, but if they aren't on any calls, it should show 0. Am I missing something here, I'm sure it's really obvious. With thanks Mike Mike

RE: [Asterisk-Users] call return?

For me this worked straight out of the box with [EMAIL PROTECTED] 0.3 Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com

[Asterisk-Users] UPS for Asterisk

is minimal, the probability of HD failure is significantly reduced. P.S. Power regulation is not needed, only protection against instantaneous power loss. Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01

[Asterisk-Users] Problems transferring calls - HELP!

accounts to our VoIP provider's * server, rather than our own, everything worked fine. Please help if you can, this is baking my noodle!!! Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL

[Asterisk-Users] Problems transferring calls - Part 2!

, even though 202 is hearing music). At the moment I can only transfer trunk calls through the parking system, which is a pain to teach people about... I'm really stumped on this one. Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02

RE: [Asterisk-Users] DIDs anywhere but here?

We have DID's in 5 Australian cities for $5 per month. Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message

[Asterisk-Users] Transferring calls on Asterisk with X-Lite

to do with the Reinvite status of the SIP phones? With Thanks Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry HillsN.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com

RE: [Asterisk-Users] On Hold music

files The command is modprobe ztdummy More information at: http://www.voip-info.org/tiki-print.php?page=Asterisk+timer+ztdummy Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED

[Asterisk-Users] Asterisk@Home systems

I am having trouble setting up Meetme with this CD. I have the latest which was posted on sourceforge about 2-3 days ago. It seems to come with meetme 8200 and 8201 rooms, but I am getting invalid messages. Can anyone help. The Meetme.conf is: conf = 8200 conf = 8201 The extensions are: exten

[Asterisk-Users] Limit outgoing trunk calls

the dialplan or the iax.conf? Thanks Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry HillsN.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message