Re: [Asterisk-Users] Booting * from CF

2005-01-04 Thread Andy Powell
On 02/01/2005 at 11:21 Michael Graves wrote: Hi All, I've read J.R. Richardson's paper Create an Embedded Asterisk Server which outlines making a Debian server that boots from a compressed disc image on a CF card. I'm really interested in this as I want my * server to be more like an appliance

RE: [Asterisk-Users] solid-state asterisk pbx?

2005-02-17 Thread Andy Powell
On 16/02/2005 at 09:00 Michael Graves wrote: Andy Powell has prepared a CF image at www.automated.it/asterisk. I have been able to get this booted on a testbed system. Sadly, I'm a Linux newbie and not skilled at command line administration, thus I'm stuck at the moment. I can get the existing

Re: [Asterisk-Users] People detected as fax machines

2004-01-15 Thread Andy Powell
If you don't have a fax connected to * then create and exten: exten = fax,1,Goto(day,s,1) I had the same today... :/ Andy *** REPLY SEPARATOR *** On 15/01/2004 at 16:41 Iain Stevenson wrote: A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17

Re: [Asterisk-Users] Junk calls from FWD numbers

2004-01-29 Thread Andy Powell
*** REPLY SEPARATOR *** On 27/01/2004 at 15:55 Chris Albertson wrote: My Asterisk server registers two FWD numbers. On average I get about one call a day from someone calling from an FWD number and leaving a pointless, under 10 second message. It's easy to see who these

Re: [Asterisk-Users] Introducing Firefly

2004-01-30 Thread Andy Powell
Hi, I downloaded this the other day and finally got it to stop crashing. It appears that any response from asterisk that implies an error (for example dialing a non-existant number, using the wrong password, selecting a codec that you've configured a local * not to use etc) resulted in a crash.

[Asterisk-Users] Code Hosting...

2004-02-04 Thread Andy Powell
lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently

Re: [Asterisk-Users] Code Hosting...

2004-02-04 Thread Andy Powell
Isn't this what the asterisk-addons directory was created for? This is where the MySQL code was relegated after it became legally unfavorable to put it in the CVS main branches. JT The code in question was actively denied entry into CVS (asterisk core or addons) Andy

[Asterisk-Users] Snom 100 + H.323

2004-02-20 Thread Andy Powell
Hi, can anyone give me any pointers as to how I should configure a snom 100 (with h.323 firmware) to use h.323 between it and *. How can I check that my h.323 install is ok too.. If i do: ASTERISK*CLI h.323 show tokens ASTERISK*CLI h.323 show codecs I get no info or anything back, if I turn

Re: [Asterisk-Users] OT: SNOM and TAPI

2004-02-23 Thread Andy Powell
Snom TAPI integration is a joke... Andy *** REPLY SEPARATOR *** On 22/02/2004 at 21:47 Peer Oliver schmidt wrote: Hi, anyone here running SNOM phones with TAPI integration with Outlook? Any other hardware phone with some TAPI integration? rgds pos

Re: [Asterisk-Users] OT: SNOM and TAPI

2004-02-24 Thread Andy Powell
which it will dial... Andy *** REPLY SEPARATOR *** On 23/02/2004 at 17:26 Peer Oliver schmidt wrote: Andy Powell wrote: Snom TAPI integration is a joke... Would you mind elaborating a bit on this? Is the future implemented, but does not work, or is it not implemented

Re: [Asterisk-Users] European Caller ID

2004-03-14 Thread Andy Powell
Take a look at http://www.ainslie.org.uk/callerid/cli_faq.htm Lots of info there Andy *** REPLY SEPARATOR *** On 14/03/2004 at 11:45 randulo wrote: Can anyone ell me if they've had experience on the continent with caller ID on analog POTS lines? Here in France, we

Re: [Asterisk-Users] MySQL Dynamic Extensions

2004-03-15 Thread Andy Powell
You could take a look at http://andreasotto.net/asterisk/ and modify that to suit Andy *** REPLY SEPARATOR *** On 15/03/2004 at 16:46 Tony Wasson wrote: Darren Nay wrote: Hello All, I am just looking into Asterisk as a viable voicemail solution for our phone

[Asterisk-Users] Mantis - closing feature request when feature no added

2004-03-21 Thread Andy Powell
Ok, so I've re-reported a feature request http://bugs.digium.com/bug_view_page.php?bug_id=0001265 because http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9 was closed for no apparent reason. Is it now policy to simply close off feature requests when they haven't been added? If it

Re: [Asterisk-Users] Newbie question on soft phones with SIP and *

2003-06-06 Thread Andy Powell
Tielman, You can take a look at the quick and dirty guide I'm slowly putting together if you like... http://www.automated.it/guidetoasterisk.htm I'd appreciate any feedback you have on it.. and if it helped Andy *** REPLY SEPARATOR *** On 06/06/2003 at 14:17 Tielman

Re: [Asterisk-Users] Newbie question on soft phones with SIP and *

2003-06-06 Thread Andy Powell
On 06/06/2003 at 17:36 Patrick wrote: Excellent stuff Andy. It was quite a disappointment that the document stopped before explaining ..errr everything :) Look forward to learn how to setup one-way conference and music on hold. Thanks for the guide so far. Regards, Patrick Glad it was of use

Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?

2003-05-29 Thread Andy Powell
Nathan, Get in touch with www.provu.co.uk ask to speak to Tim, and tell him you heard from me (Andy Powell) that they had a deal running where you could get Snom 100's for 140 gbp... HTH Andy *** REPLY SEPARATOR *** On 29/05/2003 at 12:44 nathan wrote: Hi All, What

Re: [Asterisk-Users] Call Transfer Problem

2003-06-04 Thread Andy Powell
Sorry, I might be being stupid, but I don't see what the problem is. Following your example, 1. Secretary calls someone for the Boss 2. Other caller answers, Secretary asks other end to wait. 3. Secretary presses the flash button (or recall or whatever it's called on the phone) 4. Secretary

Re: [Asterisk-Users] Getting netmeeting to work with Asterisk

2003-06-05 Thread Andy Powell
I've played with modifying the extensions.conf and h323.conf but don't have things right. I keep getting a message on the console: ERROR[376849]: File chan_h323.c, Line 974 (setup_incoming_call): Call from user 'Simon' rejected due to no default context However I am unsure what this really

Re: [Asterisk-Users] Music on hold, Call Parking, etc

2003-05-14 Thread Andy Powell
Ok, so are you pressing # then hearing the word 'transfer' and dialing the exten to transfer to? Andy *** REPLY SEPARATOR *** On 14/05/2003 at 11:34 Derek Beaumont wrote: I am using a regular analog phone. Derek, What are you using to place the call? Snom Phone? Cisco

Re: [Asterisk-Users] Newbie : i try and test to use asterisk

2003-06-11 Thread Andy Powell
Hi, You need to change your settings in X-lite: Display name : roseau user name : 1000 --- this is wrong! authorization user : same as user name Password : Domain/Realme : 192.168.0.2 SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty to: user name : roseau (That should match the

[Asterisk-Users] Monitor application

2003-06-12 Thread Andy Powell
Hi, I've had a search through the archives and didn't find much. Is anyone using the Monitor application? I have it working but there is a really big drawback. The files are always called the same thing, which means if I make 2 calls one after the other the first recording is lost. I half

Re: [Asterisk-Users] Monitor application

2003-06-12 Thread Andy Powell
Ahh, wonderful thanks... Andy On 12/06/2003 at 13:35 Pertti Pikkarainen wrote: Check http://www.loligo.com/asterisk/current/extensions.conf and find macro called macro-record-on There is at least one way described ( author is John Todd ). --Pertti

Re: [Asterisk-Users] Voicemail notification

2003-06-12 Thread Andy Powell
Ok, thanks for the clarification Shame it still doesn;t work for me :( maybe it only works with US phones... anyone in Europe got this working? Andy *** REPLY SEPARATOR *** On 11/06/2003 at 21:55 Tilghman Lesher wrote: On Wednesday 11 June 2003 19:10, Andy Powell wrote

Re: [Asterisk-Users] Asterisk switch = statement

2003-06-13 Thread Andy Powell
So is that one switch statement per installation or one per context. ie can i have multiple switch statements each one applicable to a different section in extensions.conf Andy On 13/06/2003 at 13:28 Martin Pycko wrote: The idea of switch is for every box to know what it can reach locally.

Re: [Asterisk-Users] CallerID forward???

2003-06-13 Thread Andy Powell
Derek, exten = 400,1,SetCallerID(${CALLERIDNUM}) You can use ${CALLERID} ${CALLERIDNAME} ${CALLERIDNUM} Andy On 13/06/2003 at 16:18 Derek Beaumont wrote: I don't understand how or where I would use setcallerid. I have tried to do exten=400,1,Setcallerid,asreceived but that doesn't seem to

[Asterisk-Users] Whoooaaa!!! Feaky - but in a good way

2003-06-15 Thread Andy Powell
Ok, this has really freaked me out, but in a good way - sort of.. I've made no changes at all to my system, save messing with ADSI. However this has nothing to do with ADSI. The thing is all of a sudden my DECT phones have started reporting caller id, and not just the number, the name too!

RE: [Asterisk-Users] Whoooaaa!!! Feaky - but in a good way

2003-06-16 Thread Andy Powell
On 16/06/2003 at 10:26 DUSTIN WILDES wrote: If this is through your Telco, they may have turned on the Callerid-Name field along with your number. I had mine turn on the Callerid-Name field for us. No, not from my teleco, this is from * via the TDM card to the DECT phones that's why it

Re: [Asterisk-Users] Test System?

2003-06-17 Thread Andy Powell
On 17/06/2003 at 10:23 Rushowr wrote: Is it possible to set up Asterisk without any of the cards? I'm interested in setting it up for the company I work for, but I would like to set it up and see how difficult it will be before I start having the company spend a chunk on equipment. Yes, you can

Re: [Asterisk-Users] festival error

2003-06-19 Thread Andy Powell
You have looked at festival.conf right? What's your exten line line here's one of mine: exten = 1021,1,Festival(mary had a little lamb) Note the lack of quotes hth Andy *** REPLY SEPARATOR *** On 19/06/2003 at 13:57 Chad Sawyer wrote: I followed the directions I found in

Re: [Asterisk-Users] [HS] results testing asterisk with ISDN BRI look for help to understand configuring SIP with asterisk

2003-06-20 Thread Andy Powell
Hi, I don't understand what i have to make and set to communicate with external telephons SIP (Sjphone, X-lite, MS messenger ...) Must i have a SIP provider subscription, how to integrate this subscription with asterisk Do you mean internally i.e. Sjphone, X-lite, MS messenger phones on your

Re: [Asterisk-Users] Manager interface, again

2003-06-20 Thread Andy Powell
This is odd because I email all my users voicemail out and the ones that don't clear the voicemail on the phone still get stutter tones. I had to inform them of what to do, and then mass delete their voicemail to get the stutter tone to stop. One user had almost 50 messages waiting. I had this

RE: [Asterisk-Users] Manager interface, again

2003-06-20 Thread Andy Powell
On 20/06/2003 at 14:45 Wade Weppler wrote: Same here. E-mail and MWI/Stutter tone work fine together. if that attaching the file or just sending a messages without a file attached..? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] asteisk, sip NAT

2003-06-22 Thread Andy Powell
Andy, your update is http://www.automated.it/guidetoasterisk.htm isn't it ? yes, same place, just added some extra notes in there (they should be obvious) HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] snom 100 and GSM codec

2003-06-25 Thread Andy Powell
Anybody have the latest word on Snom's development? Last I had heard, they were still working on compatibility with GSM. Firmware version 1.15e, which is what my Snom 100 automatically updated itself to, does not work with GSM. -Tilghman I'm using 1.15u and it's a little better. Snom are aware

Re: [Asterisk-Users] Asterisk hardphone

2003-06-25 Thread Andy Powell
You can try Snom or Cisco... Or get a TDM card and use an analog phone... Andy *** REPLY SEPARATOR *** On 25/06/2003 at 15:44 Chris wrote: I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware

Re: [Asterisk-Users] X100P and PSTN caller id

2003-06-26 Thread Andy Powell
Mmm... I don;'t know what else to try, I've had callerid turned on here but it doesn't work at all... :( Andy *** REPLY SEPARATOR *** On 26/06/2003 at 13:02 Dan wrote: There is nobody with an X100P in Europe having this issue related to the PSTN Caller ID? Please help!

Re: [Asterisk-Users] Asterisk - first impressions

2003-06-26 Thread Andy Powell
I tell you what, just relpy to every message with the word remove rather than actually reading the instructions. *** REPLY SEPARATOR *** On 26/06/2003 at 13:30 cisb wrote: REMOVE - Original Message - From: Peter Zeltins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent:

Re: [Asterisk-Users] X100P and PSTN caller id

2003-06-26 Thread Andy Powell
Well, I can't get it to work with an MD110 PBX over here either (.nl). It probably should work, but I never found how. Tried several options as suggested by this list though.. -- From what I can gather the caller id in NL is similar to Denmark, it's just a series of DTMF tones send down the

Re: [Asterisk-Users] Important: PSTN access-number for Dutch gateway changed

2003-06-26 Thread Andy Powell
Oliver, can you clarify how the gateways is supposed to be used, I've tried calling the number from a PSTN line, the call is answered and i get dialtone, I then try to dial my iaxtel number and just get told that it's an invalid extension.. the 'error' occurs after dialing 17001 of my iaxtel

Re: [Asterisk-Users] Voicemail issue

2003-06-27 Thread Andy Powell
Dan, The first question is : is your voicemail in the default location or have you moved it to another disk? if you do this you need to update the vm system link in the /var/spool/asterisk directory eg: vm - /home/asterisk/voicemail/default/ using ln -s new path vm also make sure * has the

Re: [Asterisk-Users] Can I disable musiconhold for agents

2003-06-27 Thread Andy Powell
You could create a simple moh class that played a silent mp3 as a very low rate, or even the occasional beepthen just use setmusiconhold,newclass hth Andy On 27/06/2003 at 13:10 Derek Beaumont wrote: I was playing with the agent application to see if I could get it to work. Everything

Re: [Asterisk-Users] modprobe ? for TDM40B

2003-06-27 Thread Andy Powell
The X100P is modprobe wcfxo The TDM40B is modprobe wcfxs Andy *** REPLY SEPARATOR *** On 27/06/2003 at 16:07 Steven P. Donegan wrote: What is the module name for the TDM40B - I received my X100P and TDM40B today (thanks Digium). ___

RE: [Asterisk-Users] Minimum budget question ...

2003-06-30 Thread Andy Powell
Tim, a good comprehensive answer to the question...certainly gave me a few things to think about. I do have a few questions though, since I'm in Europe. Has anyone in Europe set up something equivalent to what Tim suggested? What sort of prices did it work out at? How did you solve the

Re: [Asterisk-Users] Asterisk Sacrifice?

2003-07-04 Thread Andy Powell
Hi, You just have to be a little patient... try http://www.automated.it/guidetoasterisk.htm as a start, it might at least get you going with sip based stuff. I don't like to particularly push the guide specifically because it's mine. I'd rather you got it by recommendation... but hey,

Re: [Asterisk-Users] switch = priority in the dialplan.. (probably an issue for Mark)

2003-07-04 Thread Andy Powell
WipeOut, IIRC the qualify=yes directive in your iax.conf definition for the switch causes * to check to see if the host is alive. Andy Is there a way to get asterisk to verify that the remote host is in fact available before attempting the switch so that if it is unavailable the local

Re: [Asterisk-Users] Asterisk and VMWare

2003-07-07 Thread Andy Powell
Hi Dan, For a totally unrelated reason I did this today. * runs fine here under VMware, athough I haven't stressed it at all. Andy *** REPLY SEPARATOR *** On 07/07/2003 at 19:07 Dan wrote: Hi, There is any experience using Asterisk with VMWare? I think about installing a

Re: [Asterisk-Users] Problem with the Internet LineJACK ISA card...

2003-08-07 Thread Andy Powell
You also appear to have a big problem with your clock... unless you are from the future.. in which case how are Glaxo stocks doing? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] [OT] unsubscribe

2003-08-07 Thread Andy Powell
Steve I have to say that some listserv's do allow this .. at least he didn't reply to 20 messages with REMOVE in them Andy On 07/08/2003 at 10:10 Steve Meyers wrote: On Thu, 2003-08-07 at 10:01, Justin Carlson wrote: unsubscribe Has anyone ever been on a mailing list where you could

Re: [Asterisk-Users] Ansering machine/voicemail detect?

2003-08-07 Thread Andy Powell
Garry, yes this is possible although it would end up being quite convoluted. Essentially you could have a cron job that monitors your voicemail directory, or use the perl manager interface to check the status. Once it has been established that you have message(s) submit a .call file to dial

Re: [Asterisk-Users] CallerID, DECT phones and ATA

2003-08-09 Thread Andy Powell
of seconds when you pick-up the phone, to know that you have a voice message waiting. BR, Dan - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 08, 2003 2:34 PM Subject: Re: [Asterisk-Users] CallerID, DECT phones and ATA Hi Dan, I use

RE: [Asterisk-Users] Running Asterisk behind NAT?

2003-08-14 Thread Andy Powell
Is this not just a case of a new entry in sip.conf EXTERNIP = external IP with the code for the contact header modified to use it (if present). Then the external firewall is set to forward the rtp and 5060 to * .. I know many people either have sip aware firewalls (as i do) or their * box

Re: [Asterisk-Users] FXO mode

2003-08-14 Thread Andy Powell
Errm, no... does that mean you'll personally check to see if my line is busy or not ;P will try it now... Andy *** REPLY SEPARATOR *** On 14/08/2003 at 09:58 Martin Pycko wrote: Did you try BUSYDETECT_MARTIN in asterisk/Makefile ? regards Martin On Thu, 14 Aug 2003, Andy

Re: [Asterisk-Users] FXO mode

2003-08-14 Thread Andy Powell
FCC mode is for the US. CTR21 is for Europe - you even pasted the info in your message! Exactly, the question really is how do you change it? modprobe wcfxo opermode=1 HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] list proposal

2003-08-14 Thread Andy Powell
I was pondering on this question, and have to agree, splitting mailing list just means yet another list to join (since there may one day be something relavant) and filter locally. What might appear to be a good solution is a privately run newsgroup on a digium server eg news.digium.com the

Re: [Asterisk-Users] Asterisk Newbie ...

2003-08-14 Thread Andy Powell
Fabia, The only numbers you should be able to dial from that config are 1945 1943 2999 and nothing else... The entry under bogon-calls (isn't it bogus calls?) should read exten = s,1,Congestion rather that using the _. ... HTH Andy *** REPLY SEPARATOR *** On 10/08/2003

[Asterisk-Users] To Switch or not to Switch... that is the question....

2003-08-14 Thread Andy Powell
Hi, when using multiple * boxes, there appear to be 2 choices as to how to go about sharing cards and dialplans 1) using switch 2) using dial fail fall-through ie exten = 1,1, dial(xyz) exten = 1,2, dial(otherpbx/xyz) As i see it switch could end up being recurrsive resulting in a wild ooc

Re: [Asterisk-Users] New SIP Phone

2003-08-14 Thread Andy Powell
It's just a proxy service like fwd it will work with asterisk... The phones they are selling with the deal are Grandstreams. It's very likely that they just have been preloaded with their settings, and probably point to their own tftp server. simply create fake dns entries and a static route

Re: [Asterisk-Users] FXO mode

2003-08-14 Thread Andy Powell
On 13/08/2003 at 17:46 Dave Cotton wrote: in the file wcfxo.c the following structure is initialised as below which would suggest that FCC is wrong for France or pretty well all of Europe. errm, FCC mode is for the US. CTR21 is for Europe - you even pasted the info in your message! See below

Re: [Asterisk-Users] New SIP Phone

2003-08-14 Thread Andy Powell
In fact the is is not required, see below: On 06/08/2003 at 15:50 Michael Robertson wrote: The phones are completely preconfigured, but not locked in any way to the SIPphone service. Owners are free to change any settings they want. -- MR Andy Powell wrote: Hi, Might seem an obvious

Re: [Asterisk-Users] FXO mode

2003-08-14 Thread Andy Powell
___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users *** REPLY SEPARATOR *** On 14/08/2003 at 20:48 Richard Scobie wrote: Andy Powell wrote: FCC mode is for the US. CTR21 is for Europe - you even pasted the info

Re: [Asterisk-Users] FXO mode

2003-08-14 Thread Andy Powell
, Andy Powell wrote: Hi Dave, I have a similar problem, I tried using busydetect and busycount but calls kept being dropped at random intervals. It didn't seem to matter what i set the busycount to. I guess it's a case of deciding which is more important... You can also limit the length

Re: [Asterisk-Users] Running Asterisk behind NAT?

2003-08-14 Thread Andy Powell
I think this is a good idea but at least for FWD users can't they just use the FWD proxy that is designed to handle clients behind NAT with no special software on the client. The ones that allows even Windows Messenger to work behind NAT. Sadly this doesn't work otherwise they'd all be using

Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?

2003-08-14 Thread Andy Powell
Hi Mark, Short of taking my board out of my * box is there any way to check what revision of the TDM400P I have? It was purchased in May of this year. Is the pricing likely to be the same or similar to the add-on FXS ports? Does this also mean that as we'd be able to get away with not using

Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?

2003-08-14 Thread Andy Powell
Mark, if the capability for line reversal detection is in the hardware (X100P) then does this mean that the detection of DTMF style caller-id as used in the following countries would ber trivial? or am I hoping too much... Finland, Denmark, Iceland, the Netherlands,India, Belgium, Sweden,

Re: [Asterisk-Users] CallerID, DECT phones and ATA

2003-08-14 Thread Andy Powell
Hi Dan, I use panasonic DECT phones, when plugged into a TDM20B (2 port FXS card from Digium) I get caller id passed through (name AND number) although i can't get callerid via the pstn at the moment (located in nl) i do get it for VoIP calls. Plus when a pstn call comes in and there is no

Re: [Asterisk-Users] Can I runAsterisk remotely from telnet session?

2003-08-15 Thread Andy Powell
Personally I'd use ssh rather than telnet Andy *** REPLY SEPARATOR *** On 15/08/2003 at 12:21 Steve Lane wrote: I am having problems trying to run asterisk from a telnet session. I am able to su to root and the command asterisk does not work. Any ideas why this may be

Re: [Asterisk-Users] CALLERIDNUM not passed over?

2004-06-03 Thread Andy Powell
This one came up a week or so ago on list... please check the archives before posting. use 's' before the CALLERIDNUM ie exten = 999,2,VoicemailMain(s${CALLERIDNUM}) Andy On 03/06/2004 at 14:41 Reto Stauss wrote: When a user dials 999 he is always asked for the mailbox and has to enter

RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.

2004-06-03 Thread Andy Powell
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian D'Arcy Sent: Tuesday, June 01, 2004 2:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec. Hello all, I'm going to tackle learning C this week, and

Re: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.

2004-06-05 Thread Andy Powell
On 04/06/2004 at 14:36 James W. Brinkerhoff wrote: On Thursday 03 June 2004 07:05 pm, Andy Powell wrote: chan_btp Hi Brian, You might also like to take a look at chan_btp and the btp daemon which allows the use of bluetooth devices to change routing. Since any old linux box that can handle

Re: [Asterisk-Users] dialplan experts needed

2004-06-08 Thread Andy Powell
Matthew, Dial works on a fall thru principle. Thus: exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) should suit your purpose (not taking into account vm), to add another exten just add it on the dial 'list': exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) exten

Fwd: Re: [Asterisk-Users] dialplan experts needed

2004-06-08 Thread Andy Powell
*** BEGIN FORWARDED MESSAGE *** On 07/06/2004 at 23:34 Andy Powell [EMAIL PROTECTED] wrote: From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Tue, 08 Jun 2004 14:54:33 +0200 Subject: Fwd: Re: [Asterisk-Users] dialplan experts needed Sorry misread your message, you want

Re: [Asterisk-Users] dialplan experts needed

2004-06-08 Thread Andy Powell
On 08/06/2004 at 11:15 John Fraizer wrote: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup ...should be That's why

Re: [Asterisk-Users] New version of DIAX (0.9.8a) available nowfor free download

2004-06-10 Thread Andy Powell
On 10/06/2004 at 09:04 Dan wrote: Hi, - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] Cool. It is posible to use the GSM phone as a DIAX headset ? At least there is posible to transmit audio using Bluetooth. Unfortunately not, because the GSM phone does not

Re: [Asterisk-Users] New version of DIAX (0.9.8a) available nowfor free download

2004-06-10 Thread Andy Powell
Hi Dan On 10/06/2004 at 14:01 Dan wrote: Hi Andy, - Original Message - From: Andy Powell [EMAIL PROTECTED] Any chance of getting this to work with Nokia phones Dan? No chance unfortunately.. Nokia does not support the extended AT commands set needed to control phone keyboard

Re: [Asterisk-Users] How to get the Called id with AGI

2004-06-10 Thread Andy Powell
On 10/06/2004 at 14:40 Angel Diaz wrote: Hi all, Is there a way to get the called id (the B number) with AGI perl ? I know how to get the caller id which is working fine and is just below: code snip Thanks in advance, Angel. use: $exten = $input{'extension'}; to get the extension

Re: [Asterisk-Users] Dyn Exten

2004-06-14 Thread Andy Powell
On 14/06/2004 at 14:53 Jose R. Ortiz Ubarri wrote: Best mailling list support I've ever read!!! Thanks a lot for your help. Yes, unfortunately there are a couple of people on the list who will a) tell you whatever you are doing is wrong and that they know better b) but not actually offer any

RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just Software?

2004-06-17 Thread Andy Powell
On 16/06/2004 at 22:53 Jay Milk wrote: You're correct -- I believe I pointed out in my original post that there is a $200+ difference between a cordless Cisco with/without software. And that's plain ridiculous. Plus, the phone alone isn't worth $500 in hardware -- so we're obviously dealing

Re: [Asterisk-Users] looking for newbie resources

2004-07-04 Thread Andy Powell
On 04/07/2004 at 14:53 Steven M. Sawczyn wrote: Hi, I am very interested in VOIP and telephony in general, although admittedly, I don't know much about the theories and protocols behind it. Having also an interest in Linux, I was really excited to come upon Asterisk. I would really like to

Re: [Asterisk-Users] looking for newbie resources

2004-07-05 Thread Andy Powell
Hi Hank, Working on updating it, and perhaps splitting it into more than one page Andy On 04/07/2004 at 17:52 hank smith wrote: hello andy is your user guide updated? - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 04, 2004 5:24 PM

Re: [Asterisk-Users] Question about Cisco IP Phone 7960

2004-07-08 Thread Andy Powell
On 08/07/2004 at 08:21 Hall, Eric M. wrote: I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf

Re: [Asterisk-Users] Small Linux Distro

2004-07-08 Thread Andy Powell
On 08/07/2004 at 18:41 Philipp von Klitzing wrote: and you'll find a link to the Asterisk Live! CD-ROM. If you have a moment I guess the list (and certainly me) would be interested to hear about your experiences with this. :-) Awww c'mon, it's only 29mb download it and try it for yourself I'd

RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

2004-07-09 Thread Andy Powell
On 08/07/2004 at 22:19 usedcanon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an

RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net

2004-07-09 Thread Andy Powell
On 09/07/2004 at 13:25 Chris Bond wrote: On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. Install identd and allow TCP port 113 inbound access and it'll work - if you

RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

2004-07-11 Thread Andy Powell
On 11/07/2004 at 08:42 Paul Mahler wrote: You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run.

RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

2004-07-11 Thread Andy Powell
On 11/07/2004 at 12:31 Paul Mahler wrote: The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. WRONG! This doesn't have ANYTHING TO DO WITH ASTERISK.

Re: [Asterisk-Users] Please ignore my last message...

2004-07-11 Thread Andy Powell
On 11/07/2004 at 20:00 Steven Sokol wrote: Please forgive me for sending that last message to the wrong list. It was supposed to go to the Dev list. Sorry, Steven LOL, for me at least - this message arrived before whatever message you accidentally sent... :D Andy

RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

2004-07-12 Thread Andy Powell
On 11/07/2004 at 18:11 Paul Mahler wrote: Well, this is certainly getting exciting. Andy, I took your advice and re-read the RFP. Andy--I don't think you are a Sorry, I was sleeping when these new emails came in I've read the other responses which seem to make it pretty clear.. and

Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

2004-07-12 Thread Andy Powell
I don't think we should let these misunderstandings judge the quality of Paul's Asterisk book. Even authors need to learn now and then :-) Can I just point out that the reason I said what I said (see, I can't write) was because Paul steadfastly refused to believe what we were saying, rather

Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

2004-07-13 Thread Andy Powell
On 13/07/2004 at 11:48 Martin List-Petersen wrote: I can see the point of the discussion somewhere, but let's take it the other way around (comments though mail): On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote: You have not shown us ANY example yet for which this facility is *NEEDED*.

[Asterisk-Users] WARNING: Deprecated incominglimit and outgoinglimit

2004-07-13 Thread Andy Powell
For those that don't read every line of source code here's something I found out today... Deprecated incominglimit and outgoinglimit Incominglimit = number of calls the local extension can originate to Asterisk. Outgoinglimit = number of calls Asterisk will terminate to local

Re: [Asterisk-Users] Parking renamed to feature in 7/17/04 CVS

2004-07-18 Thread Andy Powell
On 17/07/2004 at 20:25 Josh Roberson wrote: Seth Remington wrote: I just updated from CVS and noticed that Mark has renamed all of the parking related files (parking.conf, parking.h, res_parking.c) to features.conf, features.h, res_features.c respectively. The CVS log mentions that this is in

[Asterisk-Users] SIP Registration issues

2004-07-20 Thread Andy Powell
Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now

[Asterisk-Users] FYI: Perl module for Cisco 79x0 phones...

2003-09-07 Thread Andy Powell
hi, not sure is anyone is aware, but I found a perl module that makes interfacing with a cisco 79x0 phone a breeze http://www.cpan.org/modules/by-module/Cisco/ Though it might be of some use... Andy ___ Asterisk-Users mailing list [EMAIL

Re: [Asterisk-Users] Using a Cisco 7960G

2003-09-08 Thread Andy Powell
Sean, The dealer is talking utter bol**cks... You simple need the SIP firmware from Cisco and a TFTP server running on a pc. I have a Cisco 7940 G running happily with an * box. Not only that but I'm able to happily use the XML interfacing on the phone for things like a phone directory and

Re: [Asterisk-Users] (no subject)

2003-09-13 Thread Andy Powell
From what I see this *IS* a problem with the CVS code... as a quick fix I suggest using the zaptel code from august 18th 2003 since that is known to work (I'm using it after having the same problems as you) It's kinda strange if this isn;t regarded as a bug, as Digium have then EOL'd some of

Re: [Asterisk-Users] (no subject)

2003-09-13 Thread Andy Powell
Yep, it probably will not work with your motherboard. You might try setting -DNO_CALIBRATION in the Makefile, then running 'make clean all install' and trying again (this has worked for some people). Failing that, try it with a different motherboard. -Tilghman This is a CODE issue not a

Re: [Asterisk-Users] (no subject)

2003-09-13 Thread Andy Powell
It's kinda strange if this isn;t regarded as a bug, as Digium have then EOL'd some of their cards and not told anyone, while continuing to sell them... Compare revision E to revision C of the card. Revision C is no longer being sold by Digium. This may be true, however, they were being sold

Re: [Asterisk-Users] error

2003-10-21 Thread Andy Powell
Your clock is wonky sync with an ntp server or set the time on your machine... Andy *** REPLY SEPARATOR *** On 21/10/2003 at 15:03 Chris Albertson wrote: --- Ron Fallara [EMAIL PROTECTED] wrote: NOTICE[1192484144]: File sched.c, Line 209 (sched_settime): Request to

Re: [Asterisk-Users] Asterisk 1.0.1 Too many open files

2004-12-11 Thread Andy Powell
On 09/12/2004 at 09:22 Eric wrote: Hi Sean, Thanks for your reply, but that wasn't exactly what I was getting at. I don't need to increase the system's imposed limit on the number of open files. I'm more concerned to see if anyone has run across a memory or fd leak in asterisk that sucks them

Re: [Asterisk-Users] Call routing based upon callerID

2004-03-29 Thread Andy Powell
John, This is referenced as the anti ex-girlfriend feature... example: exten = s/12345678,1,congestion exten = s/24681012,1,Dial(SIP/phone2) exten = s,1,Dial(SIP/phone1,30) also check page 31 of the handbook... hth Andy *** REPLY SEPARATOR *** On 29/03/2004 at 20:34

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