RE: [Asterisk-Users] solid-state asterisk pbx?
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 image running, but have not been able to get ssh working, change passwords, load my configs to the CF, etc. If there's someone on-list who could assist in this regard I'd gladly share my experience moving my production server to be CF based. Michael lo, If you are using dhcp for the test box then login and type dhcpcd then to start ssh... sshd to change your password... passwd You need to copy the password files back to the cf so that it'll be copied back at boot. To mount the 3rd partition (where the configs live): mount /dev/hda3 /mnt/cfgs remember to umount it when you're done. There are some example configs on hda3 (hint rename the examples folder to just astlive - AFTER you edit the network stuff etc :D) quick command reference: cd = cd cp = copy rm = delete ls = dir mv = move/rename longer command reference: http://docsrv.sco.com/DOS_others/Going_from_dO_to_u1.html HTH Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Booting * from CF
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 than a PC. However, the paper is only an outline and some of the processes of pruning the installation down to a minimum are beyond my Linux skills. http://www.automated.it/asterisk/asterisk-cf.htm That being the case, and CF cards being bigger and cheaper every month, does anyone have an experience building a server that simply boots from a 512 MB or 1 GB CF card? Is that big enough? It really depends on what you want to do, are you thinking of storing voicemail on the CF? The above link is to install onto a 64mb CF card (can be used on bigger cards too if required) Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.1 Too many open files
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 all up. There should be no reason that I hit my limit of open files on this machine. Restarting asterisk immediately solved the problem, so I'm leaning towards a leak, however, I didn't have the opportunity, in the moment, to check and see how many files and what type were open. - Eric I'm pretty sure that it's a leak, if I recount a problem I have (had) when trying to register with FWD is should make it obvious. About a week or two ago I started having problems with registering with FWD using SIP, the request was sent but there was never a reply. Indeed a traceroute showed a problem at peer1.net (this is still the case). I noticed that after a few hours I was getting the same errors as you. A restart of asterisk cured the problem temporarily until a few hours later, when it reappeared. incidentally I 'fixed' the issue by using an iax2 connection to fwd instead... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email from Brekeke Announcing their RTP Proxy
Is it April 1st already, where did the year go Andy On 25/09/2004 at 01:47 SeshKanuri wrote: Dear Valued OnDO users, OnDO PBX v1.3 now supports 100 concurrent calls Brekeke is excited to announce our new OnDO PBX v1.3 with increased concurrent call capacity that is 4 times greater than the current release version OnDO PBX v1.2. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
On 25/09/2004 at 14:31 Arik Funke wrote: Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine and then copy everything to the flash device.) Cheers, Arik You could start buy downloading my .iso (29mb bootable ) and use that as a basisis for your system. I've already modified it for a CF card based system. Essentially it depends what sort of interface to the pstn you want. E1/T1 and analog should work fine with my cd - but I've not built it for use with CAPI or the QuadBRI cards... you can grab it at http://www.automated.it/asterisk/ It's not v1 of * but I am trying to find the time to update to a newer CVS version, however I will only do that once I'm happy running that particular version myself... HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] dial '0' for outside line and get a dialtone...
On 17/09/2004 at 12:21 Pawlowski Julian wrote:
I'd like to create the following: a user picks up the phone
(gets a dial tone), dials '0' for an 'outside' line, gets a
second (different?) dialtone, and is able to enter an
external phone number.
Klaus-Peter Junghanns has something like this on his page:
http://83.137.99.170/jn/relaunch/asterisk/page19.html
It didn't work for me correctly so I changed a lot to fit it to my
dialplan. Give it a try for you...
Regards,
Julian Pawlowski
All seems a little OTT for me, why don't you just have:
exten = 0,1,DISA(no-password,mydialout)
[mydialout]
exten = _XX.,1,Dial(ZAP/g1/${EXTEN})
you could, if you wanted, add an _00X. or _0XX (depends on local number lengths)
to be able to dial straight out...
seems a lot simpler than the macro above... although you aren't getting a different
tone...
AIMHO
Andy
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Re: [Asterisk-Users] GSM phones, bluetooth and general happiness
On 23/09/2004 at 13:36 Joe Antkowiak wrote: There are quite a number of positive (for end users) implications of doing this too... just think about all those cell providers that offer unlimited mobile to mobile calls, and then all those unlimited LD packages from landline and voip providers. This has huge potential for people who use their cell phones alot... Not to mention the fact that you wont be microwaving your brain... :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On 09/09/2004 at 18:48 Josh Roberson wrote: I wrote cepstral regarding this at the beginning of the week, thought it might be relevant to post the reply: Thanks for contacting us. Our Linux package is off the site right now because we are releasing a new version, 3.02, next week. This is an incremental release. The major update of this version is a new Linux SDK. Please check back with us in 6-7 days and we should have what you're looking for. We appreciate your patience. -Craig Now hopefully, they'll hold up to it and release the new Linux SDK in a week or so... -twisted This *may* be related to my original app_cepstral that can't be integrated into CVS because of the licencing. bkw had a chat with them, iirc about making parts gpl, to solve this 'issue'.. perhaps they've done it (are doing it)... only time will tell Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
On 07/09/2004 at 23:57 Benjamin on Asterisk Mailing Lists wrote: On Tue, 07 Sep 2004 08:14:57 -0500, Brian Capouch [EMAIL PROTECTED] wrote: If you have a Linux laptop with you, then in fact the SIP devices can be configured to hide behind it. The laptop can then run an instance of asterisk that connects to the home asterisk server, Like I said: I run Asterisk on my Powerbook to do IAX to my company's Asterisk server. Keep in mind though that you don't need to have a Linux notebook to do this. A Powerbook running MacOSX runs Asterisk just fine. This may not be much of an issue for the Linux geeks and techies on the list, but if you have to send sales people and other non-tech folks on business trips and give them something to connect, then probably a Powerbook running OSX will be an easier choice since they get to keep their native MS-Office. At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT Easicom - Andy Powell
On 02/09/2004 at 10:08 Andrew Newton wrote: Hi, I have been looking for info on * and the BT Easicom 1000 without much luck when i found a post to this list from Andy Powell saying that he had the phone working quite well. Before i go buy a shedload of these things I would like to know what problems/sucesses people have had with these phones and * in the UK. What they can/cant do with * Also does anyone know of any good ADSI Scripting resources/tutorials? Many thanks Andrew Newton Hi Andrew, A couple of things for you to be aware of with the BT Easicom 1000 1. It's designed to be plugged into the Master socket, so if you are going to use it with * and a tdm card then you're going to need an adaptor with a ring capacitor in it - without this the phone wont actually ring it's 'bell' on an incoming call (although it will be ringing, if you pick up it will answer) 2. ADSI sucks, or at least the implementation on the BT easicom does. For example while in voicemail you'll get nasty screeching as the commands are sent to udate the display. What I ended up doing was turning ADSI on on the fxs port it was connected to, programming my menus, then turning it off again. I haven't got the extended features working as yet (too much to do too little time) HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incoming caller doesn't hear rining.
On 29/07/2004 at 15:49 Johan wrote: A very helpful person just sorted the problem out. Apparently, changing the incoming dial in extensions.conf to Tr solved the problem. Thanks ...but your caller will get a ringing tone even if your phone number is engaged... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Successfully Using $135 Avaya sip phone
Brian Elton wrote: The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns 481 extension does not exist. Anyone willing to help me figure out why? what do you mean : mailbox=context (or contect) this should be [EMAIL PROTECTED] or mailbox=number to use the default context. eg: [EMAIL PROTECTED] where sales is a context in voicemail.conf Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration issues
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 it appears that asterisk gets totally confused and tries to register with itself! Anyone got any ideas? Thanks Andy 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.nixhelp.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: sip:[EMAIL PROTECTED];tag=as72c0d7da To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 192.168.1.2:5060 Sip read: REGISTER sip:sip.nixhelp.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: sip:[EMAIL PROTECTED];tag=as72c0d7da To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.2 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: sip:[EMAIL PROTECTED];tag=as72c0d7da To: sip:[EMAIL PROTECTED];tag=as72c0d7da Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] ontent-Length: 0 to 192.168.1.2:5060 Jul 20 23:46:40 NOTICE[81930]: chan_sip.c:7320 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.2' Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sip read: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: sip:[EMAIL PROTECTED];tag=as72c0d7da To: sip:[EMAIL PROTECTED];tag=as72c0d7da Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 10 headers, 0 lines -- Got SIP response 403 Forbidden back from 192.168.1.2 Destroying call '[EMAIL PROTECTED]' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parking renamed to feature in 7/17/04 CVS
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 preparation for some more (possibly post 1.0) feature additions. The header file still #define(s) _PARKING_H though so let the confusion ensue ;) Time to update the wiki. -Seth Actually, no, that was fixed also. -twisted Excellent! can't you do it so that each time you grab a new version from CVS it uses a random filename for each and every config, just to make sure.. possibly even using the wrong filename for the wrong configs... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
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*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Now .. the problem is, that every hardware phone, every softphone etc. actually might need a different configuration, some IAX, some SIP, some one codec, some other codecs (now that we are talk asterisk). It will get quite problematic to get all solutions under one account without breaking one or the other. Yes, this is a problem I''d forsee... but ignoring that for one moment :P Imagine that asterisk accepts multiple registrations for a single entry in sip.conf ([myphone]) simply adding each to an internal variable: The first phone registers: WHO_I_DIAL = sip:[EMAIL PROTECTED] then joe comes along and also registers a line on his phone WHO_I_DIAL = sip/[EMAIL PROTECTED]sip/[EMAIL PROTECTED] now when I execute a dial, asterisk internally replaces the occurrence of myphone with the WHO_I_DIAL variable: eg: Dial(SIP/myphone,120) becomes (internally) Dial(WHO_I_DIAL,120) In essence DIAL sees nothing different at all and doesn;t need to be changed because the internal reference SIP/myphone actually = the content of WHO_I_DIAL So what we affectively achieve is: Dial(sip/[EMAIL PROTECTED]sip/[EMAIL PROTECTED],120) Which is what people have been saying everyone should do... but this process becomes automatic, which is a feature that people want. I'm pretty sure you'd do this with an array rather than a string, but I think it explains the theory behind it all. Of course I've ignored the issue with different configs required for different SIP devices (eg DTMFMODE=), but that artistic license ;) I may have explained it badly, so let me know Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING: Deprecated incominglimit and outgoinglimit
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 extension. End of Life for these commands announced**, please use setgroup and checkgroup, that will also be helpful with cross channels. There is an example on how to do this at Asterisk cmd SetGroup. It's from the viewpoint of the Asterisk PBX, not from the local extension. The CLI command sip show inuse will show the current status. The outgoinglimit is currently disabled in the source code of the SIP channel. -- ** someone is kidding here, right? Announced? I think not... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
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 address all the points and give most of the info you need...(do I need to add to it?) I couldn't for the life of me remember the name (it was late) and Andres reminded us all that it's called Parallel Forking - it's by far the best feature of SIP and nearly, but not quite, negates the NAT problems. The reason i've been so adamant about this, is that I use it every day... my * box and 2 of my phones register with a local sip proxy for the same sip address... I use this just incase my * box dies, since it's my development box too and I'm always mesing with it. good candidate for a beginner's book on *, but if you send my your address, I'll send you a copy on me. :-) Or some Ninja assasin... ;) Perhaps you could also sign it :D (not the Ninja assasin ;) ) Andy, I'm in your hands. I was too late... I took the liberty of getting some sleep... appologies. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
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 than investigating it.ie His response was more like: You're wrong, I'm right. rather than: Oh... maybe there's something I'm not aware of. I shall investigate immediately. I'll admit techies always argue over stuff like this, primarily because they don't want to be seen to not know something.. anyho.. I'd consider the discussion of the existance of forking closed and proven and we can now begin arguing over why it would/wouldn't be a good idea to include this behaviour in * ;) Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
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. The unique SIP registration is how * finds a UNIQUE endpoint. Sorry, but this is irrelavant... SIP allows multiple endpoints to register with the same account details and will all ring when called. The fact that the rtp stream goes to the first endpoint to pick up (and respond) is what's important ie, if multiple devices are registered with the same account they will *all* be 'spoken' to... Asterisk currently does not support this behaviour. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. No, the idea is to get asterisk to act like a real sip proxy. The dialplan solution is a poor hack. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. again, irrelavant - the whole beauty of the way SIP works is that I can add to the list of phones that get called by simply registering more phones with the same details. I don't need my users to mess with or make a support call to add to the dial plan. They can add and remove themselves. I'd also suggest adding something like registrationlimit=1 for those that do not want to support multiple client registrations, I'd also like to see the implementation of the q parameter... I'm all for this modification to SIP, although I'd probably want to see DTMF callerid implemented first :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
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. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. WRONG! If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. This is TOTAL rubbish .. you clearly have no idea how SIP works I think I'll skip your book Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please ignore my last message...
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
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 overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. I second that, I think a more reasonably priced book in PDF fromat would have been better. Umar. Late last year I was approached by a publisher asking if I would be interested in writing an asterisk book. I said a polite no (after some discussion) for a number of reasons: 1. Precisely what the author of this book is experiencing. Being bitchslapped by the asterisk community, for no apparent good reason. Since very few of the people on this list have actually read the book this early critism and mud slinging appears unfounded. Let's face it - the biggest failing of asterisk is it's lack of documentation. Sure there are guides, documentation projects.. but all of these rely on people giving up their free time... and since we don't have much of that, progress is slow. Anything that helps document asterisk and how to get it set up can't be all that bad. 2. I hand't heard of the publisher before, and a google search didn't turn up the most favourable links. 3. Asterisk changes day by day.. If I'd gone with it the book would have been out by now and (aside from being bitchslapped) I'd probably immediately have had to start a 2nd edition.. I'm not a writer... I can't even spell properly. I don't know what the author was offered, but if it was just 15% then perhaps the deal I was offered wasn't as bad as I thought... At $49 it is quite expensive, however, when funds allow I'll more than likely buy a copy out of interest - I consider myself fairly a well seasoned asterisk person, but hey it might teach me something too... I'm prepared to give it a chance. All IMHO of course... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
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 play about with your username it'll probably work too. Kind Regards, Chris Bond Identd is NOT required Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about Cisco IP Phone 7960
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 Any ideas? Again sorry this is off topic Make sure you changed ALL the configs to point to the CORRECT image file.. OS79XX.TXT should contain P0S3-07-1-00 and your SIPmac or SIPDefault.cnf should contain image_version: P0S3-07-1-00 iirc the default in OS79XX.TXT is the unsigned image... HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small Linux Distro
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 be interested in anyone who has done this and if it worked ok with their hardware (or not) and what the system config was... Rgds Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for newbie resources
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 Subject: Re: [Asterisk-Users] looking for newbie resources 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 learn more about Asterisk and VOIP in general and am wondering if anyone could suggest some beginner resources? Of course I've found that the best way to learn something is to just dive in and try it, but I don't think I'm ready to tackle installing Asterisk yet. In which case, http://www.automated.it/asterisk/ You'll find a link there for my Asterisk Live! CD (it's a test version, but feedback so far has been favourable) I'm running Slackware Linux on a machine which at the moment, is just hosting mail. In addition, I have accounts with both Vonage and Broadvoice. My idea is to set up a mini PBX here at home using both VOIP providers as my main lines and using my LAN to connect a few extensions. Might this be a good way to start learning, or am I way off track? Again, I am very new to this, so any info/resources/suggestions greatly appreciated. You could also try http://www.automated.it/guidetoasterisk.htm to get you going... The wiki has useful info too http://www.voip-info.org Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for newbie resources
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 learn more about Asterisk and VOIP in general and am wondering if anyone could suggest some beginner resources? Of course I've found that the best way to learn something is to just dive in and try it, but I don't think I'm ready to tackle installing Asterisk yet. In which case, http://www.automated.it/asterisk/ You'll find a link there for my Asterisk Live! CD (it's a test version, but feedback so far has been favourable) I'm running Slackware Linux on a machine which at the moment, is just hosting mail. In addition, I have accounts with both Vonage and Broadvoice. My idea is to set up a mini PBX here at home using both VOIP providers as my main lines and using my LAN to connect a few extensions. Might this be a good way to start learning, or am I way off track? Again, I am very new to this, so any info/resources/suggestions greatly appreciated. You could also try http://www.automated.it/guidetoasterisk.htm to get you going... The wiki has useful info too http://www.voip-info.org Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just Software?
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 with GREED here. My knee-jerk response to such business tactics always has been to do it better and cheaper. Six years ago, I was talking to IT personel in industry X. There were two established mainframe solutions in that industry serving 80% of the market, costing $50K-$75K start-up cost per location, plus $1K+ per seat. Never mind the $10K-$15K monthly maintenance cost. Never mind that everyone had to be able to work a terminal with a lovely amber on black, text-based GUI. snip for brevity I think you're missing the point. When you develop hardware or software you need to recoup the cost of development (the period in which you aren't selling anything, so not making any money). Now Cisco has it's fingers in many pies so they aren't going to suffer to much from that now, but they do have to fund development. Secondly, Cisco don't really care if their phones are out of your price range, they are typically sold as part of a solution costing 10's of 1000's or 100's of 1000's of USD/GBP/EUR and (most probably) with big discounts. Thirdly, If I make a device at a cost of $5 and sell it for $500, some people will buy it, up to the point where someone builds a similar device and sells it for $150 ...You have a choice. companies are not charities, they do this to make money. This is what we call capitalism. I don't want to dig at your business, and this isn't intended to but.. what you did is look at what was already on offer and it's costs, how it worked etc and built a cheaper solution. The reason you could do this is because you had the exposure to the 'system' as was.. i.e. You looked at it and said 'I can do that cheaper' but without that original system you probably wouldn't have. One final point... There are some companies that have this weird feeling that anything under a certain amount must be cheap and nasty and not work properly. These people are fools imho, but they do exist...and they wont buy an cheap phone, they'll buy an expensive phone, regardless of it's ability... as we've seen recently some governments will even buy helicopters that can't fly in fog or where it's sandy for silly money... Now I feel dirty... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dyn Exten
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 help at all. it's best to just ignore them. The good news is that the ratio is pretty low... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New version of DIAX (0.9.8a) available nowfor free download
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 support Audio Gateway profile (just Headset profile). It can connect only with the headset. ..but.. you can use the Bluetooth headset for DIAX and the GSM phone as CallerID/Dialer. .. and all this even when the computer screen is locked. Best regards, Dan Any chance of getting this to work with Nokia phones Dan? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New version of DIAX (0.9.8a) available nowfor free download
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 and display. This is one of the reasons I like Ericsson;-) Best regards, Dan Ok, but can I still used my BT headset and a BT dongle on the PC to speak? I'm thinking it's a bit easier to carry the headset about for answering calls. For dialing I'm happy to pick up a proper phone or if I'm at the PC just use the DIAX interface. Possible? Thanks Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to get the Called id with AGI
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 called.
Andy
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Re: [Asterisk-Users] dialplan experts needed
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 = 555,3,Dial(SIP/3000,30) voicemail should be positioned at (exten + 101) for busy - I'd stick noop's in to allow the hangup before the next exten = 555,1,Dial(SIP/1000,30) exten = 555,2,NOOP exten = 555,3,Dial(SIP/2000,30) exten = 555,4,NOOP exten = 555,5,Dial(SIP/3000,30) exten = 555,6,NOOP exten = 555,102,VoiceMail2(u3278) exten = 555,103,Hangup exten = 555,104,VoiceMail2(u3278) exten = 555,105,Hangup exten = 555,106,VoiceMail2(u3278) exten = 555,107,Hangup this wont allow the dial of 2000 or 3000 if 1000 is busy, it would go to vm. Ok, that's a bit of explaination, here's what you are prolly interested in exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) exten = 555,3,VoiceMail2(u3278) exten = 555,4,Hangup exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup And you probably want call waiting turned off... HTH Andy *** REPLY SEPARATOR *** On 07/06/2004 at 23:34 Matthew Simpson wrote: In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: 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 That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: Re: [Asterisk-Users] dialplan experts needed
Pah! my fingers are getting in the way today: exten = 555,1,Dial(SIP/1000,30) exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,203,VoiceMail2(u3278) exten = 555,204,Hangup Andy *** 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 it to dial the next when it's BUSY... not if it's not answered.. Disregard my previous message and use... exten = 555,1,Dial(SIP/1000,30) exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup exten = 555,101,Dial(SIP/2000,30) exten = 555,102,VoiceMail2(u3278) exten = 555,103,Hangup exten = 555,202,VoiceMail2(u3278) exten = 555,203,Hangup I've made the asumption (even if that is the 'mother of all F***ups') that if it's not answered it should just go to vm. Andy *** 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 14:47:32 +0200 Subject: Re: [Asterisk-Users] dialplan experts needed 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 = 555,3,Dial(SIP/3000,30) voicemail should be positioned at (exten + 101) for busy - I'd stick noop's in to allow the hangup before the next exten = 555,1,Dial(SIP/1000,30) exten = 555,2,NOOP exten = 555,3,Dial(SIP/2000,30) exten = 555,4,NOOP exten = 555,5,Dial(SIP/3000,30) exten = 555,6,NOOP exten = 555,102,VoiceMail2(u3278) exten = 555,103,Hangup exten = 555,104,VoiceMail2(u3278) exten = 555,105,Hangup exten = 555,106,VoiceMail2(u3278) exten = 555,107,Hangup this wont allow the dial of 2000 or 3000 if 1000 is busy, it would go to vm. Ok, that's a bit of explaination, here's what you are prolly interested in exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) exten = 555,3,VoiceMail2(u3278) exten = 555,4,Hangup exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup And you probably want call waiting turned off... HTH Andy *** REPLY SEPARATOR *** On 07/06/2004 at 23:34 Matthew Simpson wrote: In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: 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 That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** END FORWARDED MESSAGE *** *** END FORWARDED MESSAGE *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan experts needed
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 I follwed up with corrections Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
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 a bluetooth dongle can report back to a server you can have them all over the place. From what I saw you were looking at timed routing, adding bt to this might make your life a lot easier... Where can one find this chan_btp ... Google didn't turn up anything -jwb you can check it out from CVS - its called btp... :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALLERIDNUM not passed over?
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
his mailbox
number and password. As I understand this shouldn't happen because the
CALLERIDNUM is
passed over to VoicemailMain. It's annoying to have to enter the number
everytime ...
The voice mail configuration is read from MySQL. We are using the CVS
version from a few
days ago.
Extract from extensions.conf:
exten = 999,1,wait(1)
exten = 999,2,VoicemailMain(${CALLERIDNUM})
exten = 999,3,Hangup
Any hints?
Thanks in advance
Reto
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RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
-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 start writing my first * add-on/contribution; assuming it's actually worthy of contributing once it's done.. I think I've chosen a hefty project for my first go round here... I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want this to turn into a it would be cool if it did this.., or that etc... I'm writing this to serve a very simple and basic function, and I want it to do exceedingly well at just that for starters. Please check out specs below as to how I envision it working within a dialplan environment, and also, please keep in mind this is being written to be used in a corporate environment. There are a lot of others out there with far more * experience than myself, so any constructive criticism would be most welcome as to the layout and configuration of the soon to be app_findme. Thanks! snip 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 a bluetooth dongle can report back to a server you can have them all over the place. From what I saw you were looking at timed routing, adding bt to this might make your life a lot easier... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Pascal
On 28/05/2004 at 19:58 usedcanon wrote: Hi Andy, I am most certainly interested. If you have some example code using a DB (MySQL maybe) that would be extremelly helpful. BTW, I am new to fpc(Turbo pascal, Delphi and now Kylix), does it have a linux command line IDE like the DOS version Thanks for your help Umar Sorry umar, I missed your reply in the influx of messages...just spotted it...I'll tar it up and put it on my site. I've also got an example which connects to a mysql db (which I'll include) Yes there is an ide for linux... but e.. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Pascal
On 01/06/2004 at 11:00 Umar Sear wrote: Hi Andy, Once again thanks. This should make things a lot easier for me. I am greatful. btw what is the command line to execute the freepascal ide, also do you have any other recomendations. Thanks Umar. No problem, I hope it comes in handy :D I don't actually use the IDE myself, I stick to vi and just use the command line for fpc compiles... It's worth noting that you can't compile the IDE yourself, it relies on Borland libs - which is annoying try this one: ftp://ftp.freepascal.org/pub/fpc/snapshot/linux/fp.gz http://www.freepascal.org/onechap.pdf It's not that regularly updated.. but should work HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
On 30/05/2004 at 22:10 Tilghman Lesher wrote: On Saturday 29 May 2004 16:53, Andy Powell wrote: If nobody appears to know, it's probable that they haven't done the experimentation necessary to show one result or another. If you are concerned about this behavior, then it falls to you to do the necessary tests and prove it one way or the other, for the good of the community. There's a reason it's called a community -- sometimes you have to give, instead of just take. -- Tilghman Well, there are a number of resposes to this one... 1) If the recommendation (in the bug tracker) is to turn on blocking, but know one actually knows what the effect would be, well it's not much of a recommendation is it... 2) I think I *have* given something to the community, my getting started guide seems to have helped quite a few people get going, I'm in the IRC channel happy to help when I can but sometimes *I'd* like some help. 3) I *have* been testing it myself to see the effects. you assume too much Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unblocking incoming SIP
On 30/05/2004 at 21:35 Thor Atle Rustad wrote: I have just set up my first Asterisk, and I have the basics up an running. I have set it up with two SIP phones only. I can call between them, and I can call out to FWD phones. However, on receiving calls from FWD, my Asterisk blocks the calls with the following message: May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_request: Failed to authenticate user user sip:[EMAIL PROTECTED]. Obviously, I want FWD users to be able to call me without my registering them first. Any suggestions would be appreciated. Thor Thor, this is because some oh so clever person decided that the default 'security' option for sip should be to reject anything that's not in sip.conf put : insecure=very in your fwd definition in sip.conf It was basically that everything had to authenticate... which the fwd number couldn;t because it wasn;t defined in sip.conf. Anything that's added to * that breaks how protocols work should be by default OFF not ON, but that's just IMO... Of course I find it quite funny that it's insecure=very, perhaps it should be: make-SIP-work-how-it-is-supposed-to=yes ;) Blah blah moo! Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unblocking incoming SIP
On 31/05/2004 at 10:47 Eric Wieling wrote: On Mon, 2004-05-31 at 10:16, Duane wrote: Andy Powell wrote: Anything that's added to * that breaks how protocols work should be by default OFF not ON, but that's just IMO... I agree 100%, this has been very frustrating trying to work out why Asterisk suddenly stopped accepting calls from FWD and other PSTN based services. I very much preferred the old method, if I didn't want to accept a SIP call you just don't have a matching context. The problem is that too many people had a context= in [general] and didn't realize that incoming SIP calls that didn't match anything would be accepted and sent to the context= that was specified in [genera]. which is why everywhere you look in the guides etc people say put something like: context=boguscalls in the general section, which (providing you weren't stupid enough to create a [boguscalls] section worked well... in fact I'll go as far as quoting my own guide: An important point here, if you do not have a sip aware firewall and are just using port forwarding then ensure that your context points to somewhere like invalidcalls. If you do not do this then someone could call one of your extensions direct from the Internet. If you had an FXO card in the machine, this could lead to them being able to make PSTN calls Those people that didn't realize were more than likely using a guide to set up... I still stand by the fact that this feature should have been OFF in the first place. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unblocking incoming SIP
*** REPLY SEPARATOR *** On 31/05/2004 at 11:13 Andres wrote: Thats the way we prefer it (the old way). Its nice to be able to publish a sip phone number to anybody out there(for example I can just say that my number is sip:[EMAIL PROTECTED]). When the call comes into Asterisk (from whatever SIP source), the [general] section tells it to take the call to the Autoattendant in whatever context you have defined. Otherwise we have now lost that possibility. which is another good point :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
On 29/05/2004 at 13:52 brian k. west wrote: its not really a critical issue... wonder when someone will take the time and fix it. :P bkw to you bkw_ .. it's actually quite important to some of us... a bit like DTMF callerid :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Caller ID with BT CD50
On 29/05/2004 at 19:16 Tony Hoyle wrote: Me too - the current patch could also be used to do DTMF caller ID without too much work (there isn't a line reversal in the specs for that, you just have to look for valid digits). I'll probably do some tidying up (change ukcallerid to callerid=uk as it's neater).. the zaptel side though is stable. Tony Tony, there's a bounty (although it's not much but it's better than a poke in the eye with a sharp stick) for DTMF callerid (some of us have been bitching about it for ages)... http://bugs.digium.com/bug_view_page.php?bug_id=0001265 For reference Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
On 29/05/2004 at 16:49 brian k. west wrote: Accually you can issue the cli commands via manager and get full outputs! (Most people dont know that) bkw yes you can, but you have to have blocking=yes ... and I'm still waiting for info on what the implications of doing this are.. eg if the manager session is disconnected mid transmission... etc no one appears to know... or care ... or both Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Pascal
On 27/05/2004 at 22:32 usedcanon wrote: Hi, Has anyone done any AGI scripting in pascal. I would appreciate help anyone can offer. My understandin on AGI scripting is very flaky, I am assuming whatever language is used the application needs to be compile and made executable. So if I write a script in pascal, I would compile it with something like freepascal and make it executable. Thanks Umar Sear If you are still interested, I've done an FPC unit for AGI... it's freely available,... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe with AGI scripts
On 14/05/2004 at 09:00 Olle E. Johansson wrote: Andy Powell wrote: I should point out that you don;t actually have to be *using* a ZAP channel for the background agi to work. The script starts when the first person enters, once the conference is over it;s upto the script to realize this and exit (otherwise you'll end up with lots of processes laying about) ...so are you telling us that the background AGI needs a Zaptel Timer? If not, please explain what we require to run the background stuff again, so we can document it in a better way. Yes ...it works fine on my system which has and x100p and tdm20b from sip phones analog phones, iax phones.. - and I don't think I'm particularly blessed.. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: [ISN] Voice Over IP Can Be Vulnerable To Hackers, Too
I'm sorry, but any IT Manager who looks upon Internet phoning as a relatively secure technology doesn't deserve their job.. and any IT Manager that doesn't realise that VoIP is an IP service and hence susceptible to the pestilence that threatens all networked systems should be shot where they stand Andy On 14/05/2004 at 14:57 tmpm wrote: Hope this isn't too far OT, but its relevant to us. From isn.attrition.org http://www.informationweek.com/story/showArticle.jhtml?articleID=20300851 By W. David Gardner TechWeb News May 13, 2004 As voice over IP sweeps across the high-tech landscape, many IT managers are being lulled into a dangerous complacency because they look upon Internet phoning as a relatively secure technology--not as an IP service susceptible to the same worms, viruses, and other pestilence that threatens all networked systems. With VoIP, security specialist Mark Nagiel said Thursday in an interview, we're inserting a new technology into an unsecured and unprotected environment. VoIP is essentially availability driven, not security driven, and that's the problem. But Nagiel, manager of security consulting at NEC Unified Solutions, said that there are measures that can be taken to protect voice over IP from the threats that confront Web telephoning. The first step--an obvious one, he says--is to secure existing TCP/IP networks. Nagiel is finding that the new government-required regulations--such as Sarbanes-Oxley, which stipulates improved accounting record-keeping, and HIPAA in health care--are helping IT managers because they impose security discipline across-the-board. The financial and health-care fields are getting secured very quickly, Nagiel said. snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What's in ${EXTEN} ? Why does voicemail prompt for an extension?
On 14/05/2004 at 11:47 Paul Mahler wrote:
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
${EXTEN} in your example contains 99 ... you want to use ${CALLERIDNUM}
Andy
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Re: [Asterisk-Users] X100P and TDM400P non-USA Caller ID
Finland, Denmark, Iceland, Sweden, the Netherlands, Belgium, Brazil, Saudi Arabia, Uruguay,India all use DTMF So, logically the DTMF solution would be attacked first... but then I do have a bias.. :D Andy *** REPLY SEPARATOR *** On 14/05/2004 at 20:32 Senad Jordanovic wrote: I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA countries. How many of you are prepared to do this? Can you nominate your country needing this this feature! Regards, SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: [ISN] Voice Over IP Can Be Vulnerable To Hackers, Too
I'd probably shoot him too.. ;) Andy. On 14/05/2004 at 13:13 George Pajari wrote: I'm sorry, but any IT Manager who looks upon Internet phoning as a relatively secure technology doesn't deserve their job And what about security specialist Mark Nagil who was quoted (http://www.informationweek.com/story/showArticle.jhtml?articleID=20300851) as saying: The traditional voice model utilized PBXs, which were stable and secure? g. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
re: [Asterisk-Users] Fwd: [ISN] Voice Over IP Can Be Vulnerable To Hackers, Too
Mitnik is an asshole who used his friends for his own gain... 2600 hertz used to get operator mode captain crunch whistle generated 2600 hertz tone.. doesn't stop Mitnik being an asshole tho... Andy *** REPLY SEPARATOR *** On 14/05/2004 at 21:02 [EMAIL PROTECTED] wrote: Folks seem to have forgotten that the original hackers were hacking stable and secure traditional PBXs with captain crunch whistles! Mitnik ran wild through PBX's and mobille networks. Let's work to set up secure VOIP, but don't let anyone kid you about the golden days when telephones were secure! (for extra points, why's the hacker mag called 2600?) tmpm [EMAIL PROTECTED] wrote: __ Hope this isn't too far OT, but its relevant to us. From isn.attrition.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy
Have you tried calling Digium sales? Andy *** REPLY SEPARATOR *** On 13/05/2004 at 15:24 [EMAIL PROTECTED] wrote: Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking to pick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at [EMAIL PROTECTED] Thanks Michael Blood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe with AGI scripts
On 13/05/2004 at 14:57 Paul Crick wrote: I've had a quick look through the mail list and wiki but haven't yet resorted to looking at the meetme source code.. I see references to a background agi script that can run if you're using Zap channels. Am I right in saying that that script runs for each channel in the conference? Or is it a one time deal, running when the conference is created? I should point out that you don;t actually have to be *using* a ZAP channel for the background agi to work. The script starts when the first person enters, once the conference is over it;s upto the script to realize this and exit (otherwise you'll end up with lots of processes laying about) The backgrounder behind my question is that I have an IVR app where the caller will dial in to the system and interact. At some point I'll want them to zero-out to a call center operator, but once that conversation is finished, return them back to the IVR system to complete their transaction. I'm thinking I have to use a meetme conference to do this (as I want the original inbound IVR call to continue after the operator conversation)? Ok, here's my quick thoughts on this. When the caller calls, put them into a conference with the background agi running. When they need to talk to an operator, get them to press 0 (for example). When they do this, generate a call file that rings an operator which when they answer puts them in the same conference. When the operator is finished they just hang up. Use MeetMeCount to determine if the operator has left A related question: What about letting 2 Zap channels talk online then continue with IVR, like in a chatline type application? I'm thinking I can't do this from within the AGI script spawned when the call arrives and have to use the manager interface to push calls around? I think you can apply the same principle outlined above for this.. HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI.pm wait_for_digit() not working for me!!!
Ok,
the first think to do is check the permissions on the conf-background.agi ..asterisk
needs to be able to run it ...
The code I've listed below works fine for me:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
%input = $AGI-ReadParse();
$soundpath = /var/lib/asterisk/sounds/;
$timeout = 10;
while(1)
{
$input = chr($AGI-wait_for_digit($timeout));
if ($input eq *)
{
$AGI-stream_file($soundpath/banana-phone-song);
}
if ($input eq 1)
{
exit 0;
}
}
*** REPLY SEPARATOR ***
On 11/05/2004 at 10:52 Atif wrote:
Hello everybody!!!
I really need your help guys, I am using the AGI mode in meetme
application,
and I want that AGI should wait for an input from the client/user i.e. a
digit and then proceed, but I have used that AGI function
agi-wait_for_digit(), but no usemy agi just passes, or ignores this
function,
where AGI should stop here and wait for the input
.my extension in my dialplan.
exten = 21,1,answer
exten = 21,2,meetme(21|pb)
..and here is my AGI...
#!/usr/bin/perl -w
#use strict;
$aginame=conf-background.agi;
use File::Copy cp;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$char=0;
#while(1)
{
#$AGI-exec('WaitExten','25000');
#$char = $AGI-receive_char('600');
$char=chr($AGI-wait_for_digit('600'));
print STDERR input form rec char : $char\n;
if($char eq *)
{
print STDERR Dialing your number\n;
$srcfile=/tmp/mycall;
$dstfile=/var/spool/asterisk/outgoing/mycall;
open(MYCALL,$srcfile) || die Cant't open file :$srcfile
$!\n;
print MYCALL Channel:IAX2/bali:[EMAIL PROTECTED]/[EMAIL PROTECTED];
print MYCALL MaxRetries:2\n;
print MYCALL RetryTime:60\n;
print MYCALL WaitTime:30\n;
print MYCALL Context:atif\n;
print MYCALL Extension:22\n;
print MYCALL Priority:1\n;
close MYCALL;
# cp($srcfile,$dstfile);
print STDERR dialing complete...\n;
}
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Re: [Asterisk-Users] Asterisk Rhetorical Systems
hehehehhe Yes I know I use cepstral.. I wrote app_cepstral... (bkw messed with it too) Andy *** REPLY SEPARATOR *** On 10/05/2004 at 08:06 Eric Wieling wrote: On Mon, 2004-05-10 at 05:37, Andy Powell wrote: I'd love to hear how you get on Ben, but I get the feeling that Rhetorical's software prices are out of the reach of most people here. I think integration of this would be a very good move tho. Quite frankly Rhetoricals tts is the best I've heard so far. Try www.cepstral.com They have a wide range of voices, runs on both Linux and Windows, and is US$30 for the non-development version. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Rhetorical Systems
I'd love to hear how you get on Ben, but I get the feeling that Rhetorical's software prices are out of the reach of most people here. I think integration of this would be a very good move tho. Quite frankly Rhetoricals tts is the best I've heard so far. Andy *** REPLY SEPARATOR *** On 10/05/2004 at 11:27 Ben Merrills wrote: Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit I've not yet figured our how to change the voice, and I'm sure the quality could be improved on (in which case, does anyone know how to switch to the Male British voice?) Any advice or guidance here would be greatly appreciated. Kind Regards, Ben Merrills Internet Applications Developer Griffin Internet www.griffin.com http://www.griffin.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_sms - rocks!
Ok, I just thought I'd publicly pat Adrian Kennard (revk) on the back for this application. This is an excellent contribution and gets my vote for app of the year. For those that aren't aware app_sms allows you to send/receive fixed line sms messages from asterisk. ( you can take a look at a quick page showing this http://www.automated.it/asterisk/sms.html ).. I should point out that that this works over IAX2 as well, this means IAXy's with sms capable phones work, and also messaging between asterisk boxes also works. In fact, although I'm guessing, if there was a sip phone that did fixed line sms I bet that would work too. So big pat on the back for Adrian Kennard (revk), Thanks for all the effort and for sharing! Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] default caller id from X100P
In /etc/asterisk/zapata.conf before the channel=x (where x is the channel assigned to the FXO port) put: callerid=PSTN Call 1234567 You will need to restart * for this change to take effect Andy *** REPLY SEPARATOR *** On 09/04/2004 at 10:56 Victor Perez wrote: Is there a way to set default caller id info to pass to * when the telco does not provide it? Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] res_motv: Request for Comment
Just curious, but why does it strike you as such a bad idea? Especially if it was disabled by default. I can understand you not wanting your system security or your personal privacy compromised, but I think it would be great to have it in place for: A) Manual activation for those who want automated updates. B) CLI execution for occasional comparison to the current set. As a completely seperate application I'd probably not object (like zttool), I just don't think that this should be any part of *. Bearing in mind the number of outages that occur, the number of potential failures when connecting via the net. It's not just a privacy issue it's a functionality issue... if I manage 20 servers (same version) I don't need all 20 of them to say what are the fixes for this .. I just want 1, and I want to do it manually... the other thing is that there isn't an easy way to check what version you are running, after updating from cvs the version doesn't always change (annoying at the best of times)... ...but the core of my 'problem' is software that calls home. Perhaps it should be possible to flag the request with a token indicating that you don't want to be part of the survey, and you don't want your IP/host information stored. A --anon option, if you will. I would expect the --anon option to be the default, any communication to be encrypted, and the db secure... I can only wait until we see M$ like activation implemented... oh the joy... I am going to guess that you're joking. I just don't see that happening. Mark and the team at Digium seem dedicated to open source and to the Asterisk community. His asking for comments on this idea is a pretty good indicator of his concern for the community's opinions. I was being facecious It would be much better just to have the information present on either the Digium site or some other location. I see little point in wasting your valuable time doing something like this when there are so many outstanding issues and feature requests that could offer more. Perhaps Mark's time could be spent on other things, but I would still like to see Digium offer this option -- perhaps one of the other developers could head up the effort? True, but just take a look at the bug tracker, feature requests are getting rejected because we don't have time, we're too busy fixing bugs Just putting the current version information up on a web page is nice, but it doesn't allow me to automatically query the system and discover known issues and possible solutions. I think that, for service providers that could very well be a critical next step. Several of my clients made the decision to go with RedHat based on their update service (for which they gladly paid). I was actually thinking more along the lines of enter your version details on a web form and you get the updates and fixes info based on that, it's then only 2 minutes coding away for a simple perl (or other) app to automagically retrieve the info if you require. This makes it purely voluntary and you have to actively request the call home feature, saving us from programmer coding errors which could accidentally enable it by default during bug fixes etc. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: Sasquatch, the Loch Ness Monster, UFOs and...
On 08/04/2004 at 10:00 John Todd wrote: Any Day Now(tm). Wasim has fallen off the face of the Earth, but I've seen with my own two eyes a working copy of the Iaxy from Digium, so this holds promise. My request for a 1u 24-port IAX-based box that takes Digium daughterboards (FXO or FXS) generated some interest when a show of hands was asked for at the VON show... Bob Knight seemed to have an interest and some time on his hands. ;-) Nope, Wasim is alive and kicking and I have the demo iax phone here, it's currently doing a tour of Europe and then is off to the USA... (complete with mouse cheese ;) ). I also have an IAXy here and can confirm it works very well (except for not being able to turn of ADSI eerrk).. I've mentioned a couple of times that I'd like to see an equivalent of an IAXy channel bank device (much like you describe) unfortuantely this idea was pooh-poohed by people who simply had no interest in it. I'd be prepared to take a look, but telecoms electronics is not my field... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup on SIP unreachable?
This is a known issue with SIP - look at bug 207 in the bug tracker Andy *** REPLY SEPARATOR *** On 08/04/2004 at 12:37 Scott Laird wrote: I've noticed a little problem with my setup. I've been using a flaky version of X-Lite for testing, and it tends to crash every few phone calls. Since I'm just using it for testing, I don't really care, but it's exposed a problem: when the SIP client goes away, their calls are left in limbo. I just had to soft-hangup a multi-hour outgoing call that had belonged to my X-Lite client. Is there a way to treat SIP UNREACHABLE as a hangup? I'm running ~2 week old CVS code right now. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
I'd like to give this one 10 thumbs down. IMHO a bad idea, a nasty little bad idea.. evil, spawn of Satan. If this were implemented the first job of a new update would be to rip it out and flush it down the nearest toilet. I can only wait until we see M$ like activation implemented... oh the joy... It would be much better just to have the information present on either the Digium site or some other location. I see little point in wasting your valuable time doing something like this when there are so many outstanding issues and feature requests that could offer more. Andy *** REPLY SEPARATOR *** On 06/04/2004 at 22:31 Mark Spencer wrote: I've been considering the nature of Asterisk, its security, the bug tracker, and more... And i've come up with an interesting idea: A message of the version. The idea is that Asterisk has a compile time 32-bit unsigned int version which is incremented whenever some major new bug is fixed. When Asterisk starts up (and periodically, maybe once per day), it sends a packet with the version number to a server at Digium, along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium server replies (if it receives the packet, if not, it might get sent again in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are associated with that version of the code. In this way, an asterisk administrator could easily see if there were any major issues, critical security updates, etc, that his system might need to be updated for. Now, of course, any time you put a call home feature in, there are people who will be concerned about privacy. Clearly it will be able to be disabled, but I want to run my idea about deployment by everyone here and see if you guys had some ideas. The idea would be that *new* installs (make samples) would have the feature turned on for MAJOR level by default, and that any existing install (e.g. /etc/asterisk/sip.conf exists, but not /etc/asterisk/motv.conf) would have the file created at the next make install based upon prompting the installer. Any feedback on: a) The idea itself -- is it a good one or is it stupid? b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? Thanks! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank?
I'd take a look at the VoiceTronic cards ( http://www.voicetronix.com/hda.htm ) which can be used with * or their free software.. these cards can be configured as : 12 Loop-Start ports only. 8 Loop-Start AND 4 Station ports. 4 Loop-Start AND 8 Station ports (default configuration). 12 Station ports only. HTH Andy *** REPLY SEPARATOR *** On 07/04/2004 at 08:00 John Vogel wrote: Four or five analog lines can be done with a single computer so no channel bank is needed. If you need 6 or more than there is also the choice of using two machines and IAX. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime Lanning Sent: Tuesday, April 06, 2004 12:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Bank? quote who=Ken Hello, I'm new to Asterisk and would like to know how you could have 4 to 6 incoming analog POTS lines connecting to the Asterisk server and have 4 to 6 analog lines going out.(A T1 line is too costly). Would 2 channel banks be used? A T1 channelbank has 24 channels, so only 1 is needed. FXO channels (What you connect to the POTS lines) can be both inbound and outbound. If you are not using DID. So, you just need to find out how many concurrent calls you need to support. If you are using analog DID lines, then those are inbound only, and require FXS ports. (You supply dialtone and battery, the carrier's switch picks up your line and dials into your PBX.) Now, there are multiple ways to get the analog lines into Asterisk... o use an external gateway... POTS - SIP - Asterisk o wait until next month and get the FXO multiport cards from Digium o get a T1 card + channelbank -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
This is a fairly simple thing to do. You don;t say what type of phones you are using,
so I;ll assume SIP for the example:
Step 1:
Put
callerid=Darren 1234
for each phone definition in sip.conf, obviously replacing Darren with the user eg
Darren Nay or Joe Bloggs, then replace the 1234 with their extension number. The
format looks a little odd with the 1234 outside the quotes, but it's correct.
In your extensions.conf:
exten = *55,1,Ringing
exten = *55,2,Wait(1) ; Make the user feel like something is happening
exten = *55,3,VoiceMailMain(s${CALLERIDNUM})
The last line will not prompt for a password, if you omit the 's' it will...
HTH
Andy
*** REPLY SEPARATOR ***
On 07/04/2004 at 16:23 Darren Nay wrote:
Hey All,
We are using Asterisks as a voicemail only application, and so far all is
great. (Excellent product!)
However, I do have one question that I am hoping you might be able to help
me with.
In our asterisk application. When our customers call *55 (our dialplan
code
to check voicemail) then they are sent directly to voicemail (asterisk).
Asterisk then gives a voice prompt asking the customer to enter their
extension number (entire 10 digit telephone number in our case).
My question is. Is there a way to make asterisk aware of the calling-from
(callerID) number so that it will automatically detect the number and then
go directly to asking them to input their password.
If so, where would I make the config changes for this in the asterisk
config
files, and does anyone have an example of a similar config?
Thanks!
Darren Nay
VOIP Network Developer
Ionosphere, Inc
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk + Cisco 7920 + chan_sccp or chan_skinny
Alternatively, put it somewhere where we can all get at it :D Andy *** REPLY SEPARATOR *** On 02/04/2004 at 06:52 Raymond McKay wrote: I am using one version of their chan_sccp with a 7960, and can vouch for its functionality there. If you strike out finding an up-to-date version on the net, I can send you a tarball. I would appreciate it if you could. I was able to pull v 0.2 from a website listed in the archive but it doesn't seem to have the mods for the 7920 listed in the code yet. I'm assumning this was something put in later CVS versions but the CVS server no longer seems to be working for the site. I believe it should be small enough to email it to me off the list if you could. Send to [EMAIL PROTECTED] Thanks Raymond McKay President RAYNET Technologies LLC (860) 833-9720 http://www.raynettech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Still trying program - phone call
On 02/04/2004 at 11:17 John Chambers wrote: Andy Powell wrote: 1 Access to the PSTN - this can be done via a single X100P card (plugs into a standard phone line) or one of the sinlge port T1 cards or 4 port TDM410 cards (if you need a shedload of lines). You can also use a VoIP - PSTN gateway or gateway service (such as, but not limited to, NuFone) If you use the X100P, then as I say, a standard analogue phone line is all you need (you can add upto 3 X100P's iirc without issues). If you want to use the T1 cards then you need to get your local telco to deliver T1's to your location Actually, this is very much like one of our plans. Of course, if we are to install hardware to connect to the PSTN locally, there's little point in using a complex package like asterisk. I've done enough modem programming, including implementing SLIP and PPP from scratch, to know how to handle that. But we were hoping to use the glorious new VOIP approach, so we won't have to have a staff to babysit the special hardware. The prospect of lower costs for internet calls also gets people's attention, but supporting a modem bank is a nightmare that we're hoping to avoid. Ok, that's fine except VOIP can bet dodgy - how would you tell for example that the audio being delivered wasn't being broken up into unintelligable blips and squeeks because of bandwidth issues - particularly over the net? Personally I think the medical service should just employ more people for home visits, since seeing a person is better than just hearing an automated voice on the phone.. (you may have differing opinions, but I come from a culture of free healthcare (however bad it is at the moment)) Lots of people would agree with you there. But in modern America, this is becoming less and less feasible for most of the population. How true and how sad... :( It looks to me like you put 800 with a context of callme in your .call file... Nope; there's no 800 anything in any file that I edited. I'm assuming that it's the result of some default calculation, but I don't know yet. When in doubt blame aliens.. You could of course pay my air fare to Boston (and back) and hotel costs and I'd gladly help you out in person.. after my time at VON in Boston last year I wouldn't mind visiting again :D Yeah; wouldn't it be nice to persuade our employers to pay for this? It reminds me of advice I've often given musical friends: Don't complain about the way that people think that other musicians from far away are better than the local yokels. That way, the local audience pays to fly your friends in for parties and jam sessions, and their local audience pays to fly you there. You want to encourage this attitude; it's to everyone's advantage. Yes! OTOH, we're talking about software to do remote communications. It's probably far better if the developers are forced to do their work across the Net. If you want the software to work at a distance, it's reassuring to know that the developers know how to work at a distance. Some of us can do it remotely, but I like to visit places - and Boston isn't too unlike home (but that's beacuse of it's history :) )... /me casually increases taxes on tea and runs away... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LARGE BREASTS Handoff back to * from * via IAX?
Please don't tell me you deliberately used LARGE BREASTS as part of the subject for this... Adny *** REPLY SEPARATOR *** On 31/03/2004 at 18:16 Zot O'Connor wrote: How do I do this 1) ZAP- * - IAX(1) -- IAX(2) - DG104S -- Handset 2) No Answer on Handset 3) Back to IAX(1) 4) IAX(1) tries a cell phone 5) Still no Answer 6) Local * Voicemail. I have 1 working, and I had 4 working when there was only one box, i.e. when the handset did not answer the DG, asterisk went to the next step. Now that I have step 1 going to another server, it does not get to 4, 5 or 6, it defaults to the default mailbox on IAX(2). So how can I have IAX(2) drop the call in a way that IAX(1) keeps going, have IAX(1) timeout? Thanks! -- Zot O'Connor [EMAIL PROTECTED] White Knight Hackers, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LARGE BREASTS Handoff back to * from * via IAX?
mmm... I just wondered, since it's very likely that most people ended up deleting it
*because* of the subject line. .. so it probably wont help ... well it might...
..When you dial the IAX(2) box you have 2 choices, stop using voicemail on that
machine or makeyour dial timeout less than the ring timeout on IAX(2)
ie...
On IAX(1)
exten = _X.,1,Dial(IAX(2)-box/${EXTEN},10)
exten = _X.,2,Dial(Zap/g1/${MYCELLPHONE},10)
exten = _X.,3,VoiceMail(U12345)
On IAX(2)
exten = _X.,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20)
exten = _X.,1,VoiceMail(u123456)
etc...
so the timeout on IAX(1) is 10 and the timeout on IAX(2) is 20 ... if you call from
IAX(1) to IAX(2) then it will timeout before reaching the value of 20 and return to
IAX(1).
Potential problems:
You might not be giving the person long enough to answer on IAX(2)
Cell phone may go to voicemail, and you'd never know (timeout on that line would need
to be less than the voicemail timeout on the cell phone)
So, large breasts may have got you the answer you need... but not for the reason you
expected...
HTH
Andy
*** REPLY SEPARATOR ***
On 01/04/2004 at 16:27 Zot O'Connor wrote:
On Thu, 2004-04-01 at 04:09, Andy Powell wrote:
Please don't tell me you deliberately used LARGE BREASTS as part of
the subject for this...
I got got tired of asking questions that did not get answers while
watching people berate dead subjects or each other. The questions have
been thought out, I guess the subjects were not sexy enough.
And, unfortunately, you proved me right :)
Adny
*** REPLY SEPARATOR ***
On 31/03/2004 at 18:16 Zot O'Connor wrote:
How do I do this
1) ZAP- * - IAX(1) -- IAX(2) - DG104S -- Handset
2) No Answer on Handset
3) Back to IAX(1)
4) IAX(1) tries a cell phone
5) Still no Answer
6) Local * Voicemail.
I have 1 working, and I had 4 working when there was only one box, i.e.
when the handset did not answer the DG, asterisk went to the next step.
Now that I have step 1 going to another server, it does not get to 4, 5
or 6, it defaults to the default mailbox on IAX(2).
So how can I have IAX(2) drop the call in a way that IAX(1) keeps going,
have IAX(1) timeout?
Thanks!
--
--
Zot O'Connor [EMAIL PROTECTED]
White Knight Hackers, Inc.
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Re: [Asterisk-Users] Still trying program - phone call
John, Yes, asterisk can do that, and in fact it's very simple. The problem at the moment is your level of knowledge of asterisk, but this can be resolved... There are a number of things you need: 1 Access to the PSTN - this can be done via a single X100P card (plugs into a standard phone line) or one of the sinlge port T1 cards or 4 port TDM410 cards (if you need a shedload of lines). You can also use a VoIP - PSTN gateway or gateway service (such as, but not limited to, NuFone) If you use the X100P, then as I say, a standard analogue phone line is all you need (you can add upto 3 X100P's iirc without issues). If you want to use the T1 cards then you need to get your local telco to deliver T1's to your location The process I think you are trying to create would be something like this: Your script reads a database and generates call files for each person it needs to call. The call file contains the number to ring and the context and extention to deliver the called person to. This extension simply runs an AGI script that plays the menu and waits for user input ( various things can be done if the user does not respond eg flag a potential user in distress) - this information is then fed back to the database so that you can report on it... (or alternatively raise an immediate alarm). The AGI can be scripted in almost any language you like. Personally I think the medical service should just employ more people for home visits, since seeing a person is better than just hearing an automated voice on the phone.. (you may have differing opinions, but I come from a culture of free healthcare (however bad it is at the moment)) You are correct, the WiSIP is just a distraction, besides, from what I saw at CeBit, wait a few months and you'll have more choice. I'd suggest getting a copy of SJPhone (what we call a softphone - ie it's software not hardware) from www.sjlabs.com - it's a nice simple interface. It looks to me like you put 800 with a context of callme in your .call file... I suggest that you abide by the adage, learn to walk before you run ... You can take a look at my guide at http://www.automated.it/guidetoasterisk.htm (there are others) which may help clear up one or two points of understanding... You could of course pay my air fare to Boston (and back) and hotel costs and I'd gladly help you out in person.. after my time at VON in Boston last year I wouldn't mind visiting again :D HTH Andy *** REPLY SEPARATOR *** On 01/04/2004 at 17:35 John Chambers wrote: A while back, I asked about using Asterisk in a medical environment where the task is to write a program that connects to a phone and sends a message like: Hello Mrs. Jones. How are you doing today? Press 1 if you're OK. Press 2 if you need help. Or start talking, and your message will be passed to a person. After connecting and sending the sound file, the program would obviously need to listen for keys and voice, and do something sensible with them. Since then, I've done a bunch of installing, testing, and especially experimenting with variants of the sample.call file. So far I haven't been able to answer the question of whether what's wanted is possible. Maybe a couple of questions that we've come up with will clarify things. One is whether we can make a call to a regular land-line or cell phone, or just to VoIP-type phones. If it's possible, what do we need to know about routing? We're guessing that we need to somehow relay through some sort of IP-to-PSTN gateway, but information on this seems to be rather muddy. Asking our local telcos (and Boston has a bunch of them ;-) gets a lot of clueless responses. If we mutter the acronym VOIP, they perk up and start trying to sell us their promised VOIP phone service. But this has nothing to do with what we want to do, which is to get a *program* to make the call. This obviously implies that the connection escapes from the IP cloud and enters the PSTN cloud, but how? If we need to purchase service with some gateway provider, how do we ask for it? Actually, I've been really tempted to get a WiSIP phone, to get familiar with that. But as far as I can tell, it would just take time away from the real project, so I haven't. OTOH, if using it would make VoIP clearer to us newbies, maybe it would be a good idea. Or maybe a softphone on my Powerbook would be a better way to go. Or both? In any case, talking to a SIP phone isn't very interesting to us yet, since few people have them. A demo would have to be to the phones on people's desks or in their pockets. Meanwhile, another sort of question is how to find explanations of asterisk's many cryptic error messages. For example, after cleaning stuff out, downloading from CVS, doing a make and make install (and ignoring errors ;-), then firing up asterisk -vvvc and copying one of my test*.call files to the outgoing directory, I got: *CLI -- Attempting call on
RE: [Asterisk-Users] Call routing based upon callerID
Well, it is what he asked for, perhaps it was because I didn't do all of it for him, since I wanted him to learn rather than just copy... Let me explain: John : The scenario is that I want all calls originating from number x to be routed to a particular extension exten = s/12345678,1,congestion This means that any call with callerid of 12345678 will execute the congestion application. The application at the end can be anything at all, a Goto, a dial whatever... John: those from yy to another exten = s/24681012,1,Dial(SIP/phone2) the same as above but for 24681012 and that it run the dial application.. John: and anything else to a third. exten = s,1,Dial(SIP/phone1,30) neither of the above 2 were met, (no callerid or callerid not matching) so dial a different phone... If I've suddenly become unable to understand English then let me knowif not then hopefully this explains how to use the feature... Andy On 29/03/2004 at 20:42 Matthew B Marlowe wrote: I don't think this is what he was trying to do - And if it was, well then I'm trying to do something else. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Monday, March 29, 2004 7:16 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call routing based upon callerID 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 John F. Baird wrote: Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk ....
- Let the caller know its position in the queue (ie: you are number # in the queue, please hold and an operator will hang on you) This is not possible at the moment.. Anyone know better? Actually it is possible have a look at the bug tracker - I would give you the url but I can't get to bugs.digium.com at the moment Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk ....
Senad, I can do better than that: http://bugs.digium.com/bug_view_page.php?bug_id=214 which says that the patches have been merged into cvs :D HTH Andy *** REPLY SEPARATOR *** On 30/03/2004 at 17:00 Senad Jordanovic wrote: Andy Powell wrote: - Let the caller know its position in the queue (ie: you are number # in the queue, please hold and an operator will hang on you) This is not possible at the moment.. Anyone know better? Actually it is possible have a look at the bug tracker - I would give you the url but I can't get to bugs.digium.com at the moment Hi, Do you know which section of the bug tracker this may be in... As you know browsing mantis is quite slow, and searching is just very difficult. :) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call routing based upon callerID
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 John F. Baird wrote: Hi, I've search and though I've found a few references I have not been able to find any concrete examples of * routing a call based upon the caller ID. The scenario is that I want all calls originating from number x to be routed to a particular extension, those from yy to another and anything else to a third. Can someone please provide a reference to samples or tell me if it cannot be done easily? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mantis - closing feature request when feature no added
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 is now policy please let us know so that we can save everyone a lot of time by not bothering to add feature requests in the first place... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Dynamic Extensions
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 service. In order to use it though I will need to make extensions.conf dynamic (ie. Via MySQL). Is this possible? Sure.. Set up the database as you see documented. You can schedule this up to once a minutes using crontab. Setting something like this in your /etc/crontab should do it nicely */5 * * * * root /usr/local/sbin/update-voicemail 21 /var/log/vm.log NOTE: You can increase the frequency by using */3 or */1. */5 means every 5 minutes. */3 means every 3 minutes. Then make /usr/local/sbin/update-voicemail look like: #!/bin/bash /path/to/retrieve_extensions_from_mysql.pl /usr/sbin/asterisk -rx extensions reload Next, make the script executable #chmod +x /usr/local/sbin/update-voicemail This is not totally dynamic, but it ought to be close enough. You could make this completely dynamic using a trigger. Tony Wasson I've found the following information on this subject: http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql However, this is not a fully dynamic function. It requires me to pull the mysql database every so often (presumably via cron) and then restart asterisk after updating extensions.conf. Is it possible to setup asterisks so that extensions.conf is fully dynamic via a MySQL database? Thanks for the help!! Regards, Darren Nay [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] European Caller ID
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 currently do not have it enabled on any of out lines, but I would consider paying for it if I thought * via X100p cards would be able to detect it properly. I know for example that when people are calling us from the US or from London CID is not captured by France Télécom, because we have a free service here where you can call a number to see if you missed any calls. Are the technical specs of CID different all over? Comments, experiences please? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: SNOM and TAPI
The feature isn't really implemented.. you can install the 'driver' but you only get the ability to dial, no real status information etc... I got so fed up i wrote a little app to use http://ip of phone/dialnumber_en.htm?sip:number@ip of asterisk box, the app also allows me to click sip: uri's 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 at all? Or something else? Thanks rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: SNOM and TAPI
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 100 + H.323
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 on debug and trace I get nothing at all Can someone run through their settings for the snom h323 confid and a user entry in h323.conf Ta muchly Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Code Hosting...
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 this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there moo Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Code Hosting...
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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Introducing Firefly
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. I've only tested the IAX proto not sip or your own network. running XP with uptodate patches on a local lan. When it works it works really well, although I don;t particularly like in initial beep and end beep when i make a call (I haven't played with all the options so it may be that I can turn this off).. sound quality is good. All in all a nice little app. Are you planning on allowing other people to run your server side (like Jabber does) in their environments? If you need any further debugging info on the crashes, let me know... HTH Andy *** REPLY SEPARATOR *** On 28/01/2004 at 12:11 Adam Hart wrote: After many months of development, I'm pleased to announced Firefly - an IAX soft phone and network. The firefly softphone - free, runs under windows, allows connection to multiple networks, skinable interface, connection to firefly network, IAX2 protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw, GSM. - contact lists, selectable ringtones. download from here - http://www.virbiage.com/firefly/ The firefly network - also free, runs on an enhanced version of IAX2 (simply uses IAX2 text messages for customised part), voicemail, text messaging, online presence, ability to indicate status (available, away, NA). I believe you can connect using a standard asterisk box but you'll miss out on the extended features. The network runs on iLBC so unforunately it won't work with most IAX2 clients (unless you get * to translate) Thousands of people have used it but it's still regarded in beta, as we are still in heavy development (so expect a few bugs). It doesn't use iaxcomm as we needed our own framework to support sip, including our own jitterbuffer. If you don't wish to connect to the firefly network, click cancel when it asks you. Coming soon features SIP - in alpha, few bugs outstanding music onhold - playing mp3s while the other party is onhold fast audio - will reduce the latency by 40-50ms speex - (if anyone wants it?) Feel free to contact me on or off the list to report bugs and suggestions. I'll post everytime we release a new version (probably every week), including fixed bugs and new features Our website is http://www.virbiage.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junk calls from FWD numbers
*** 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 people are if I look in my CDR file I can see thier name and number. They seem to be new FWD users, likely who've just downloaded FWD's Xten softphone and then dial some random FWD user (me) to try it out. I wonder if these same people when they first got a POTS phone installed in thier home got out the white pages and dialed randomly asking anyone who'd answer Hi does this work? can you hear me? Question: Does everyone with an FWD number get these junk calls or am I the only lucky one? There are a number of things you can do: 1. Make sure you are not listed in the white pages (turn it off from your settings page) since FWD is a community it's pretty much accepted that if you list in the white pages you are open to receiving calls from people you don't know. Hopefully they are at least respecting your timezone settings. The FWD white pages bears no resemblance to a 'normal' white pages .. they share only a name. 2. If the calls are nusance calls then get in touch with Ed Guy and report the problem - (Don't rely on the caller id as to where the call came from.) 3. Keep in mind the reason that you don;t get calls on your pstn line with people saying Hi does this work? can you hear me? is because pstn calls are tried and tested over many years. voip doesn't have this (pstn calls don't get NAT issues) luxury. There is of course nothing you can do about people dialing random digits... HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] People detected as fax machines
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 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include = mailboxes include = day|07:55-23:00 include = night exten = t,1,Voicemail2,100 exten = t,2,Hangup [day] ; set music on hold for parked calls exten = s,1,setmusiconhold,default exten = s,2,responsetimeout,20 ; ring SIP for 20 seconds exten = s,3,Dial,sip/ciscosip/cisco1sip/cisco2sip/cisco3|20|tT ;if nobody answers tell them how to use the voicemail system. ; exten = s,4,Background,vmprompt exten = s,5,Voicemail2,100 exten = s,6,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More words for Allison
ok, how about inside outside up down server status current is and finally: Please look at bug 207 :D Andy *** REPLY SEPARATOR *** On 11/01/2004 at 19:36 John Todd wrote: Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address [EMAIL PROTECTED]. I did not include all possible symbols on a North American keyboard, as it was getting exhausting and possibly silly. calls abandons staffing average Speed of Answer Sorry, but the user's mailbox can't accept more messages. Please enter the conference call number for the conference you wish to join. fortieth fiftieth Please enter the conference pin number. That pin is invalid for this conference. [The alphabet - a through z, like ayy, bee, cee, etc.] zed space dash dot comma slash exclamation point ampersand percent at sign [we want a verbalization of the @ symbol] with plus equals left bracket right bracket open parenthesis close parenthesis pipe backslash comma period quote greater than less than chance of cloudy sunny sun turning to rainy rain partly partially mostly snowy snow scattered patchy wind windy miles per hour kilometers per hour knots per hour storm warning watch thunderstorm hail weather lightning fog foggy sleet sleeting clear clearing freezing freeze hurricane tornado severe later morning afternoon evening late early changing in the Alpha Bravo Charlie Delta Echo Foxtrot Golf Hotel India Juliet Kilo Lima Mike November Oscar Papa Quebec Romeo Sierra Tango Uniform Victor Whiskey Xray Yankee Zulu Niner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AbsoluteTimeout Users Messages
I'd be nice to be able to play a tone (or message) at AbsoluteTimeout - N where N is a number os seconds before the cut-off... a bit like pay phones (used?) to do... eg. beep beep beep beep beep click... Call terminated because you took to long explaining your probelm to the support team, goodbye. Andy *** REPLY SEPARATOR *** On 08/01/2004 at 15:05 Matteo Brancaleoni wrote: exten = T,1,Playback(yourtimehascomeahahahaha) Il gio, 2004-01-08 alle 14:26, Senad Jordanovic ha scritto: Hi, All Is there a provision for AbsoluteTimeout application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AbsoluteTimeout Users Messages
Nicolas, I'd appreciate a copy of this if possible... got a url where I can grab it? Thanks Andy *** REPLY SEPARATOR *** On 09/01/2004 at 10:43 Nicolas Gudino wrote: Andy Powell wrote: I'd be nice to be able to play a tone (or message) at AbsoluteTimeout - N where N is a number os seconds before the cut-off... a bit like pay phones (used?) to do... I have implemented an 'horrible' patch that sort of works. I'm not very good at C, and I'm new to asterisk. It makes a tone at 40, 30, 20 and 10 second before absolute-timeout. I can provide you with the patch, but its really really ugly, with lots of if/endifs. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kedpad less extension
You can use immediate=yes ;like the bat phone (old Batman) in your zapata.conf for the channel... but that means it just gets answered.. I think your problem is having the operator signal that they can take the call... otherwise when they get bag from the toilet/coffee break/ciggie break they'll find an irate customer on the other end. if they also have a pc you could write an interface to the management console to answer the call with a click... personally I prefer hardware solutions.. :D Andy *** REPLY SEPARATOR *** On 08/01/2004 at 13:58 [EMAIL PROTECTED] wrote: Does anyone know of a resource for extensions in which the server (whether asterisk or custom scripts) can trigger the phone to be answered? So for example an operator can have a headset and when a call comes through the call is automatically (through a script) connected to the headset instead of the operator having to manually answer the call. Any responses, help or ideas of a type of supplier to contact for more information would be greatly appreciated. Thanks Michael Blood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems compiling cdr_pgsql
Hi, Having installed postgresql-devel-7.4-0.3 and postgresql-libs-7.4-0.3 I'm having probs. compiling cdr_pgsql, can anyone offer any pointers as to what I might be missing? I'm hoping I've just missed out something like postgresql-wibblewobble-7.4-0.3 or something ... Below is the result of a make in the cdr source dir which may help those of you in the know thanks... Andy [EMAIL PROTECTED] cdr]# make cc -o cdr_pgsql.so cdr_pgsql.o -lpq -lz -L/usr/lib /usr/lib/gcc-lib/i386-redhat-linux/3.2/../../../crt1.o: In function `_start': /usr/lib/gcc-lib/i386-redhat-linux/3.2/../../../crt1.o(.text+0x18): undefined reference to `main' cdr_pgsql.o: In function `pgsql_log': cdr_pgsql.o(.text+0x168): undefined reference to `ast_log' cdr_pgsql.o(.text+0x192): undefined reference to `ast_log' cdr_pgsql.o(.text+0x4c1): undefined reference to `ast_log' cdr_pgsql.o(.text+0x506): undefined reference to `ast_log' cdr_pgsql.o(.text+0x535): undefined reference to `ast_cdr_disp2str' cdr_pgsql.o(.text+0x5c4): undefined reference to `ast_log' cdr_pgsql.o(.text+0x60e): undefined reference to `ast_log' cdr_pgsql.o(.text+0x65c): undefined reference to `ast_log' cdr_pgsql.o(.text+0x6ae): undefined reference to `ast_log' cdr_pgsql.o(.text+0x6d8): undefined reference to `ast_log' cdr_pgsql.o(.text+0x771): more undefined references to `ast_log' follow cdr_pgsql.o: In function `my_unload_module': cdr_pgsql.o(.text+0x988): undefined reference to `ast_cdr_unregister' cdr_pgsql.o: In function `my_load_module': cdr_pgsql.o(.text+0x9b7): undefined reference to `ast_load' cdr_pgsql.o(.text+0x9ed): undefined reference to `ast_log' cdr_pgsql.o(.text+0xa0e): undefined reference to `ast_variable_browse' cdr_pgsql.o(.text+0xa3f): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xab6): undefined reference to `ast_log' cdr_pgsql.o(.text+0xae9): undefined reference to `ast_log' cdr_pgsql.o(.text+0xb11): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xb88): undefined reference to `ast_log' cdr_pgsql.o(.text+0xbbb): undefined reference to `ast_log' cdr_pgsql.o(.text+0xbe3): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xc5a): undefined reference to `ast_log' cdr_pgsql.o(.text+0xc8d): undefined reference to `ast_log' cdr_pgsql.o(.text+0xcb5): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xd2c): undefined reference to `ast_log' cdr_pgsql.o(.text+0xd5f): undefined reference to `ast_log' cdr_pgsql.o(.text+0xd87): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xdfe): undefined reference to `ast_log' cdr_pgsql.o(.text+0xe31): undefined reference to `ast_log' cdr_pgsql.o(.text+0xe4b): undefined reference to `ast_destroy' cdr_pgsql.o(.text+0xe78): undefined reference to `ast_log' cdr_pgsql.o(.text+0xea5): undefined reference to `ast_log' cdr_pgsql.o(.text+0xedb): undefined reference to `ast_log' cdr_pgsql.o(.text+0xf08): undefined reference to `ast_log' cdr_pgsql.o(.text+0xf35): undefined reference to `ast_log' cdr_pgsql.o(.text+0xf62): more undefined references to `ast_log' follow cdr_pgsql.o: In function `my_load_module': cdr_pgsql.o(.text+0x1079): undefined reference to `ast_cdr_register' cdr_pgsql.o(.text+0x10a9): undefined reference to `ast_log' collect2: ld returned 1 exit status make: *** [cdr_pgsql.so] Error 1 [EMAIL PROTECTED] cdr]# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dutch/DTMF Caller ID
hi, since development of dtmf caller id under * is prolly going to only be done if someone stumps up the cash I've been looking for alternatives... Hoving found a number of projects which turn out to be mad prototypes or unavailable details i nearly gave up.. then I found this: http://www.artech.com.tw/html/english/ex200/ex200.htm http://www.artech.com.tw/html/english/ex200/ex200me.PDF The units are pretty cheap if i recall my conversations correctly...if anyone else in .nl interested in one of these... perhaps we could get together to reduce shipping costs... any takers? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Hans although your somewhat right I don't think its fare to ask all tourists to leave their clothes at customs and to don clogs and ride a battered old bike around the city. I also must say that from my experience its very rarely (I've never heard of it) the native Dutch that perform these crimes. You forgot constant cheese eating, complaining to the Germans about the return of bicycles, insiting that the trains are better than the UK, all while while naked or shoving a banana in some orifice Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radius on *
On 17/11/2003 at 18:39 Steve Totaro wrote: looks like critchy is especially bitchy With all his whinging, if i didn't know any better, I'd suspect he was using a 2400 baud modem... Now I'm off to reply a message and change the subject line Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : For the SQL gurus..
Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s OUCH! those times are lng! Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error
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 schedule in the past Have you looked on and around sched.c, Line 209? What's there? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
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 their cards and not told anyone, while continuing to sell them... Andy *** REPLY SEPARATOR *** On 12/09/2003 at 12:37 Jim Paraschou wrote: I have problem with a TDM40B installation. When i modprobe wcfxs the error i get is the following: /lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod /lib/modules/2.4.19-4GB/misc/wcfxs.o failed /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod wcfxs failed Does anybody know the poblem? __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
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 motherboard issue since I've happily been running a 2 port card up until a recent CVS update. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
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 in May of this year and I don;t expect a piece of hardware to have a lifespan of 3.5 months! From what I hear revision C cards are green and revision E cards are blue. It certainly also sounds like some people were getting the C version up until Digium ran out. The more i think about it the more it annoys me Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
