[Asterisk-Users] Re: Howto get voicemail $VM_ vars into externnotify script?
On Mon, 2004-10-25 at 11:09, Wilson Pickett wrote:
I am trying to slap together a script that will email2sms the details of
the voicemails left on my * box to my gsm phone. I can't figure out how
to get my script to pick up the voicemail vars like ${VM_MSGNUM},
${VM_DATE}, ${VM_MAILBOX}, ${VM_CALLERID}, ${VM_DUR}.
change the voicemail.conf to include them in the body using the emailbody
as in
emailbody=${VM_NAME} ${VM_MAILBOX}\n\nYou have received a ${VM_DUR}
message from ${VM_CALLERID}. The message was left on ${VM_DATE}.
Hi Wilson,
Thanks for your suggestion. I already have normal voicemail notification
via email working. Just as you pointed out. What I want to do besides
this is use a script configured in externnotify=/home/patrick/myapp.sh
in voicemail.conf to email this same data to a special email address
that will forward that data automatically as an sms message to my gsm
phone. This simple script doesn't even work:
$cat myapp.sh
#!/bin/sh
echo $emailbody
So either I am doing something not right or maybe these vars are not
exported (or whatever the right word is) by app_voicemail and can't be
accessed by other applications.
Any ideas?
Thanks,
Patrick
Patrick,
For every voicemail message left there is a text file placed in the
voicemail folder that has the information you are seeking. Your external
script or application good check this file and and extract the information
accordingly.
Bear in mind that the last message left will be the highest filename so for
example if you have the following files in the INBOX
msg.txt
msg0001.txt
msg0002.txt
in this case msg0002.txt would be the latest message and the message that
you are being notified about by externnotify.
I hope this information helps you solve your problem.
Umar
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[Asterisk-Users] RE: Fax detection in voip channel
usedcanon wrote:
Hi All,
Is it possible to detect an incomming fax just as it is possible with
Answer
on a Zap channel. If not do others find the possibility of this
enhancement
useful too?
Detecting that an incoming is a FAX has been present in * since its
early days.
Regards,
Steve
Steve,
Can you kindly elaborate on that. As I mention in my original post, I
understand that it is possible for calls comming in on a Zap channel, but my
tests and wiki documentation suggest that it is limited to that.
If you can shed some more light on this I would be greatful.
should something like this work ? if the incomming call is from a SIP
channel
[default]
exten = s,1,Answer
exten = s,2,Dial(IAX2/3987,40,r)
exten = s,3,Hangup
exten = fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,2,rxfax(${FAXFILE})
Thanks
Umar
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[Asterisk-Users] Fax detection in voip channel
Hi All, Is it possible to detect an incomming fax just as it is possible with Answer on a Zap channel. If not do others find the possibility of this enhancement useful too? Umar ___ 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] Fax detection in voip channel
-Original Message- From: Lee Howard [mailto:[EMAIL PROTECTED] Sent: 21 October 2004 23:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: usedcanon Subject: Re: [Asterisk-Users] Fax detection in voip channel On 2004.10.21 14:49 usedcanon wrote: Hi All, Is it possible to detect an incomming fax just as it is possible with Answer on a Zap channel. If not do others find the possibility of this enhancement useful too? Doing fax over SIP or IAX would be a frustrating effort, and a complete waste of time, IMO. See: http://www.opencall.org/faq/x47.html If you don't believe me, go ahead and actually *try* to send/receive a fax through a WAN/internet VoIP connection. You'll probably get tolerable results with SIP-fax on a LAN, but run it through a VoIP provider over the internet, and you'll have a mess, even if the codec is ULAW/ALAW What you really want is a T.38 channel driver. Lee. I understand what you are saying however there are scenarios where fax over voip works fine, I have tested (briefly) with spandsp and have done so sucessfully. as a sperate solutions we use mediatrix gateways very successfully for fax transmision over IP, our IP network is private and does not touch the internet so we can gaurantee (to some extent) bandwidth and quality. A T.38 solution would be most desirable, no doubt, but is there one ? I don't even see a mention of it anywhere. Umar ___ 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] manager interface to barge
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolás Gudiño Sent: 20 October 2004 18:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] manager interface to barge Hello, On Wed, 20 Oct 2004 09:48:43 -0600, TELUX [EMAIL PROTECTED] wrote: Can the Manager interface be used to barge my phone into an existing conversation? You need to use manager redirect and meetme. Check out my Flash Operator Panel, it lets you barge on calls. http://www.asternic.org -- Nicolás Gudiño Buenos Aires - Argentina Thats interesting, can you explain a bit more how that is done. I would like to implement something simillar without using the Flash operator. Umar ___ 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] manager interface to barge
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bill Seddon Sent: 20 October 2004 22:06 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] manager interface to barge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolás Gudiño Sent: 20 October 2004 18:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] manager interface to barge Hello, On Wed, 20 Oct 2004 09:48:43 -0600, TELUX [EMAIL PROTECTED] wrote: Can the Manager interface be used to barge my phone into an existing conversation? You need to use manager redirect and meetme. Check out my Flash Operator Panel, it lets you barge on calls. http://www.asternic.org -- Nicolás Gudiño Buenos Aires - Argentina Thats interesting, can you explain a bit more how that is done. I would like to implement something simillar without using the Flash operator. Umar Hey, Umar It seems a bit cheeky asking the guy who wrote Flash Operator Panel how to get something done so you don't have to use it. I'm sure Nicolas will reply but it might be helpful to him to learn from you why FOP doesn't work for you. If its a feature thing, maybe it a feature he can add and we all win. Bill Seddon Bill, My reason is very simple :-).. I want users to be able to do something simillar using there handsets. The reason I asked is that I am assuming that Nicholas is transferring the calls to a meetme conference, using the management api and then landing the third person in the same conference in listen only mode. The fop is gpl, so I could actually look at the code to work out exactly how it implements the barge. I am not a perl programmer (I think that's what FOP is mostly done in) so thought I ask a straight question. Hope this makes sense and Nicholas can share his knowledge (although he already has) Umar ___ 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] Anybody - please help me with this
I suggest you report it as a bug Umar -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Smith, Simon JSent: 19 October 2004 03:02To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Anybody - please help me with this Hi all, I have a problem that I cannot seem to resolve and find an answer for.I use AGI and the Perl AGI library to record wave files at various times through out a dial plan. If these recordings go over say 15 minutes or above (approx), all of a sudden the channel stops caring about DTMF escape digits and no matter what keys you press you cannot escape the recording. It is almost as if something is building up and large recordings somehow affect DTMF recognition. My max record time is 3 hours (108ms) and I am using a command similar to the following:my $x = $AGI-record_file($wavfile, 'wav', '0123456789', 108, 1); I verified that the file is still recording and getting written, but any digit I press after an extended amount of time is completely ignored. Please can anyone help me! Thanks.SS ___ 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] Free G.729 ready for download
Hi All, I consider the License fee charged by digium for G.729 as very reasonable, and hope people agree and do nothing to jeopardize this project. Right now I don't use G.729 at all, however if and when I do, I have no reason to seek an alternative to what Digium provides. At the very least I would be confident that I am in no way breaking the law, and have the satisfaction of have contributed back to the product, be it in a very small way. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Spencer Sent: 25 September 2004 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Zak Subject: Re: [Asterisk-Users] Free G.729 ready for download Regardless of whether or not you have licensed G.729 from SIPRO independently of Digium, the distribution of the codec, linked against Intel's proprietary IPP library, is clearly and totally in direct violation of the terms of the GPL. There is no room for argument on this issue. We are doing our best to mitigate the situation by removing the link to the illegal software from the list, and I ask that no more members on the list post any URL's to illegal software. Setting aside the extreme tackiness of using our own mailing list to post illegal software, I am nothing short of appalled to see an Asterisk user taking advantage of our hard work and then producing a product to circumvent the very revenue stream which makes it possible for us to do so and to offer the LEGAL licensing of G.729 to the community. Digium has worked hard to produce Free Software for building a phone system and we have released EVERYTHING we've done under GPL or other open source license, with the exception of the items we are not permitted to make available under that license. For G.729 specificially we have had to make a large investment to make that possible. The GPL DOES provide certain requirements for anyone distributing Asterisk code or derivative works, however and this unlicensed version is clearly in violation of those obligations. Why on earth would we try make the even larger investment for legal G.723.1 if people are just going to break the law and violate the GPL in order to save a few bucks? If you don't like G.729 because of the patent and licensing issues, then don't use it, but if you do want to use G.729, please use it legally, by purchasing the Digium licenses, not by breaking the terms of the GPL and putting yourself at risk of well established patents (especially if you are in a country which honors software patents, since the GPL passes that patent responsibility back to you as the end user). We are happy to make Asterisk available to the community and to continue to work hard to expand and develop the product further, but it also demands a certain level of discipline from the users at large. Before you download the free, illegal GPL-violation version of the G.729 codec, remember that in doing so you are directly jeopardizing the project at large and our ability to continue to provide these sorts of features. Mark On Sat, 25 Sep 2004, Steve Underwood wrote: Danny Zak wrote: Hello TELUX, could anybody post something more about being legaly correct using this codec and the corresponding royalty's. It is very difficult to be legally correct with this. The IP holders don't have simple programs for selling licences in small quantities. If you buy licences from Digium, they deal with the IP issues on a larger volume basis. Unless you want to deploy thousands of copies, I doubt you can find a sane legal arrangement for doing it. Regards, Steve ___ 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] American vs English
Hi Mark, Just a couple of points/requests I'd like to add. 1. Change the Commedian mail to something more generic, like Voicemail or Welcome to voicemail 2. The password prompt, just says password, would it not be better to be a bit polite and have something like, please enter your password 3. will there be any official .gsm versions of these files? 4. Is there a disclamer/license that allow these prompts to be used in a commercial setup Thanks Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Phillips Sent: 22 September 2004 23:37 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] American vs English Folks, A few people have made me aware of some omissions in my files (not my fault, they weren't in the Script from the Wiki) which I shall be tackling this weekend. Whilst I'm making the files are there any other files you want? IVR's etc. If so make sure I have a script sent by email. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ 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: Meetme
Hi all, Is there any basic information available for app_conferense? Does it suport SIP and other codecs Any installation guide Thanks Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tom Ivar Helbekkmo Sent: 24 September 2004 06:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Meetme Steve Kann [EMAIL PROTECTED] writes: This was what I wrote on the iaxclient list previously: Cool, Steve -- thanks a lot! Conference() works great for me now. :-) I've extended the description for show application thus: static char *descrip = Conference(name[/flags[/priority[/probstart[/probcont):\n Creates or joins a telephone conference. There is no configuration file;\n everything is controlled through parameters to the invocation from an\n extension context.\n name: an alphanumeric string identifying the conference\n flags: a concatenation of flag characters chosen from the following:\n M: user is a moderator, i.e. is allowed to speak\n L: user is a listener, i.e. may only listen in (default)\n T: user has a telephone, not an iaxclient; enable speex\n V: sets speex flag SPEEX_PREPROCESS_SET_VAD\n D: sets speex flag SPEEX_PREPROCESS_SET_DENOISE\n A: sets speex flag SPEEX_PREPROCESS_SET_AGC\n priority: not currently used\n probstart: sets SPEEX_PREPROCESS_SET_PROB_START value\n probcont: sets SPEEX_PREPROCESS_SET_PROB_CONTINUE value\n Returns 0 if the user exits with the '#' key, or -1 if the user hangs up.\n ; -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Free G.729 ready for download
Will this run on and AMD based machine ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Daniel Pocock Sent: 24 September 2004 17:50 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Free G.729 ready for download DISCLAIMER: This code is free (I am not charging you to use it), but you might have to pay royalty fees to the G.729 patent holders for using their algorithm. I finished this last Saturday and have had it on an Asterisk machine for 5 days without a crash, so I'm hoping that means it's safe to release into the real world. This code has also been released on the -dev list. As it is still somewhat new, I would invite anyone with feedback to forward it to me personally or to the -dev list, as I don't monitor the -users list very often. http://www.readytechnology.co.uk/open/g729 ___ 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] English vs American voice files
I am trying to download this file, keep getting page can not be displayed. Has anyone else got it on a mirror ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Phillips Sent: 20 September 2004 02:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] English vs American voice files OK, I've posted the orignal WAV files in 44.1KHZ x 16bit mono format here http://g7ltt.dyndns.org:8010/VoIP/vmukmale-wav.tgz (26MB!) Mark Mark Phillips said: Erm, didn't think of that. Stupidly I deleted the individual wav files. Not a problem though as I have the 3 master files that I recorded them all into. I'll just have to slice it up again. That'll be a few days as I've got family arriving today. Mark Linus Surguy said: I've spent the afternoon recording all the files for the English speaking VM etc. I've parked the file here http://www.g7ltt.com/VoIP/vmukmale.tgz I did it with Audacity at 44.1KHz x 16bit and thenused sox to raise the levels to -3db and then again to down sample them into 8KHz GSM files. The few that I've listened to sound fine. Hi Mark, If you're going to publish these for public use it would be great if you could make them available in two versions, both the Asterisk 'standard' .gsm format, but also either in 8KHz/8bit/alaw raw or wav and/or 32Kbit ADPCM format - these do give a noticable increase in quality for local/PSTN users of telephony applications over GSM format. Either that, or if you could make the original 44.1K 16bit masters available so others could create the alternatives. Unfortunatly *'s ability to play these cleanly seems a bit broken at the moment, but at least we'll have them for when its fixed! Linus -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Can't get ChanSpy to work
Hi Matt, There could not be a more timely response, It seems to have worked for me. added chanspy.so to the make file. thanks Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt G Sent: 14 September 2004 21:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can't get ChanSpy to work I think you have to add it to the Makefile within the apps directory Follow the instructions for Meetme2 and it should get you started available here : (http://www.areski.net/asterisk-meetme/about.php) (add chanspy.so, add commands to build it near the bottom, et al) Matt usedcanon wrote: Hi Patrick, A bit more help needed. How did you compile chanspy. I copied the chanspy.c file to asterisk/apps folder applied the muisc on hold patch, and did a make clean, make and make install. Every thing in the /asterisk/apps folder got compiled apart from chanspy. I am sure I am missing something basic, your help will be appreciated. p.s, I downloaded the patch this morning and it contained no Readme, if you have the readme from the original patch I would appreciate if you could send me it. Thanks Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Patrick J. Conroy Sent: 13 September 2004 01:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Can't get ChanSpy to work btw, was this information available in the readme or something, I am sure I looked but did not find anything. Umar, happy to help. The example in the README didn't work for me, but I may have just done something wrong. I figured it out just by testing it bunch of different ways. Patrick -- This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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't get ChanSpy to work
Hi Patrick, Nice to see you got this working. I am a bit confused as to how this works. In particular I am trying to understand, if I had say 10 active sip calls (I am particularly interested in sip), how do I specify which one I one I want to listen into. Idealy I would like to have the ability to specify the extention that I want to listen into, is that possible? I will greatly appreciate your feedback Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Patrick J. Conroy Sent: 11 September 2004 19:51 To: Oleg A. Arkhangelsky; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Can't get ChanSpy to work Oleg, Thanks very much for your help. That fixed it. Patrick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Oleg A. Arkhangelsky Sent: Saturday, September 11, 2004 3:36 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can't get ChanSpy to work Hello Patrick, Saturday, September 11, 2004, 4:03:45 AM, you wrote: PJC When I dial this extension from a SIP phone, and then make a call (which I PJC am trying to monitor) from a Zap channel, I get the following error: PJC Sep 10 20:03:35 WARNING[270359]: file.c:475 ast_openstream: File zap does PJC not exist in any format PJC Sep 10 20:03:35 WARNING[270359]: file.c:779 ast_streamfile: Unable to open PJC zap (format ULAW): No such file or directory It seems that you haven't copy files that is located in the directory called sounds of chanspy.tar.gz to /var/lib/asterisk/sounds. -- Best regards, Olegmailto:[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 -- This message has been scanned for viruses and dangerous content, and is believed to be clean. -- This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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't get ChanSpy to work
Patrick,
Thanks a lot for your response. I will give it a go in the next day or two.
btw, was this information available in the readme or something, I am sure I
looked but did not find anything.
Once again thanks
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Patrick J.
Conroy
Sent: 12 September 2004 18:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Can't get ChanSpy to work
Umar,
I am a bit confused as to how this works. In particular I am trying to
understand, if I had say 10 active sip calls (I am particularly interested
in sip), how do I specify which one I one I want to listen into.
I haven't had enough chance to test ChanSpy to know what happens when
multiple calls are going on, if you have a generic statement like:
exten = _*53,1,ChanSpy(scan)
Idealy I would like to have the ability to specify the extention
that I want
to listen into, is that possible?
That statement that I had in my original post was just for testing purposes
to get this working. The actual statement that I have in my extensions.conf
is:
exten = _*53#812X,1,ChanSpy(scan,SIP/${EXTEN:4})
This allows me to specify the extension that I want to listen on (any 4
digit SIP extension beginning with 812), and I only monitor one extension at
a time. I'm sure you could do nifty things with this to monitor any type of
extension, but this works for me.
Hope this helps,
Patrick
--
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RE: [Asterisk-Users] voicemail email messages
Why not just use wav ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of defiance Sent: 13 August 2004 16:44 To: asterisk Subject: [Asterisk-Users] voicemail email messages I am having problems with * voicemail emails. It will only mail out the gsm encoded messages. I would really like it to use the wav49 format to make it easier on my user's. I have tried specifying it with just wav49, and like wav49|gsm. It emails nothing, but if I set it to gsm|wav49 then sure enough it emails it out. According to the wiki it should send whatever is specified first. Anyone have any ideas? Chris Locke Network Administrator Stratitec 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] External MW Lamp On/Off
Hi Greg, checkout http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20voicemail.c onf From the same page externnotify Want to run an external program whenever a caller leaves a voice mail message for a user? This is where the externnotify command comes in handy. Externnotify takes a string value which is the command line you want to execute when the caller finishes leaving a message. The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox. These arguments are passed to the program that you set in the externnotify variable. END I will try and digout my script if I can. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Greg Blakely Sent: 14 August 2004 20:04 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] External MW Lamp On/Off Yes, it helps quite a bit. It shows me where Comedian Mail spawns the external app. Do you have a copy of your SIP MWI script? I may be able to use it as a starting point. Also, can you tell me what variables are passed from asterisk to the app? Thank you very much. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Umar Sear Sent: Saturday, August 14, 2004 7:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] External MW Lamp On/Off I have done something simmillar, but not the same. I send mwi notification to our softswitch (SIP). Basically I wrote a small app in pascal that sends a sip message to the softswitch. The app is called everytime a message is left or retrieved, using the extrennotify option in voicemail.conf. You could easily do something simillar, what you need to do, is write a script or app (if one does not already exist) that creates call file based on the parameters passed by externnotify. Hope this helps. Umar --- Greg Blakely [EMAIL PROTECTED] wrote: One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp when the message has been played. I understand that this has something to do with a file that is placed in /var/spool/asterisk/outgoing, but I have no idea about + what the contents of that file should be, + how Comedian mail would initiate putting the file into the outgoing queue, and + how Comedian mail would initiate putting the 'extinguish' file into the outgoing queue. Has anyone done this sort of thing already? If so, can you point me in the right direction? As I mentioned in yesterday's post, I did find a question and partial answer to this in the asterisk-users archives, but I need a bit more information before I can make it work for me. Thanks in advance for any help you can give me. ___ 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 ___ALL -NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.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 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 in india
I am a fan of Mohammad Rafi, so could do with some calls from India ;-) Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kanuri, Seshu Sent: 13 August 2004 21:14 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk in india There is one more twist to this. The broadband connection in India really sucks as it carries heavy load of Hindi Films and Hindi Film Songs. We have a few Broadband connections running from India. When the call comes from India to USA, the VOIP Broadband Phone turns into a Juke Box and all that our users in US can listen to is the Voice Overs of melodies of Lata Mangeshkar, Manna Dey and Mohammed Rafi. The VOIP Voice is critically low. Another factor is that the noise eliminates any real use of advanced codecs like G729. The Internet bandwidth is so expensive to run from India, there is no real use of hosting your Asterisk box there, Vis-a-Vis the same in USA. And offcourse you cannot legally interconnect VOIP with Analog or Digital PRI lines there. Seshu Kanuri ___ 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] 1.0 RC2 External AGI Issues
What does your dialplan look like ? I have scripts running on various versions of asterisk upto the latest checked out this morning and have seen no problems so far. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William R. Lorenz Sent: 16 August 2004 18:44 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 1.0 RC2 External AGI Issues Hi All, I'm trying to execute an external AGI script but get just the following: -- Executing AGI(SIP/xlite-2fa7, agi-test) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test -- AGI Script agi-test completed, returning 0 For whatever reason, it looks like the AGI script is just exiting without feeding the correct commands into Asterisk. I can execute the script just fine from a shell. Asterisk is running as root, and all users have +rx permissions to the script and can execute it just fine, anyways. This is the sample AGI script included with Asterisk, which I would think should work. I also tried another AGI script built with Asterisk::AGI, to no avail. Does anyone have any suggestions as to going about debugging this? -- _ __ __ ___ _| | William R. Lorenz [EMAIL PROTECTED] \ V V / '_| | http://www.ohiolinux.org/ ; Free conference and event hosting \./\./|_| |_| Linux and OSS-related topics. October 2, 2004 - Columbus, OH. ___ 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] voicemail email messages
I use just wav and that works fine. As for the smaller file size, if you use something like wav49|gsm then you are storing file in two format, so effectively that is taking up more space than just wav. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of defiance Sent: 16 August 2004 23:33 To: asterisk Subject: RE: [Asterisk-Users] voicemail email messages Doesn't work with wav either. Plus I would like the smaller file size. But if there is some way to make that work then I'm game. chris On Mon, 2004-08-16 at 17:01, usedcanon wrote: Why not just use wav ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of defiance Sent: 13 August 2004 16:44 To: asterisk Subject: [Asterisk-Users] voicemail email messages I am having problems with * voicemail emails. It will only mail out the gsm encoded messages. I would really like it to use the wav49 format to make it easier on my user's. I have tried specifying it with just wav49, and like wav49|gsm. It emails nothing, but if I set it to gsm|wav49 then sure enough it emails it out. According to the wiki it should send whatever is specified first. Anyone have any ideas? Chris Locke Network Administrator Stratitec 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 ___ 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] How to notify the user about new message using SMS
Why don't you use extern notify rather then running a script in the
dialplan.
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bartosz
Wegrzyn
Sent: 08 August 2004 20:57
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] How to notify the user about new message using
SMS
Hi,
I would like to be able to notify users about new voicemail messages
through the SMS system. I added a system command in my extensions.conf
file that sends a SMS message to the user. I am using a sendSMP.pl script
that I downloaded from the web.
It works fine when executed separately, but if executed within asterisk it
is not working. This is a part of my context:
exten = 0,1,Playback(pls-wait-connect-call)
exten = 0,2,SetCallerID(17734660101)
exten = 0,3,SetCIDName(Operator)
exten = 0,4,SetMusicOnHold(default)
exten = 0,5,Dial(SIP/[EMAIL PROTECTED],20,m)
exten = 0,6,Wait(1);
exten = 0,7,Playback(im-sorry)
exten = 0,8,Playback(nbdy-avail-to-take-call)
exten = 0,9,Playback(pls-lv-msg-will-contact)
exten = 0,10,voicemail,s
exten = 0,11,System(/scripts/sendSMS.pl -r 17083519199 -p TMOBILE -s
operator -m ${CALLERID})
exten = 0,12,Hangup
and this is the error:
-- Executing System(SIP/192.168.0.3-0891abc8, /scripts/sendSMS.pl -r
17083519199 -p TMOBILE -s operator -m Bartosz Wegrzyn 7734660101)
in new stack
/bin/sh: line 1: /scripts/sendSMS.pl -r 17083519199 -p TMOBILE -s operator
-m Bartosz: No such file or directory
The script is not working because of the characters in caller ID.
Is there any way to change that so asterisk will pass the variables
without characters.
Or do you know any other way to send SMS messages.
I tried to create the separate executable file with the command to send
sms message, but then I dont have a caller ID. Is there any way that I
could pass the $CALLERID variable to my script.
Also, the command that I execute is before the asterisk goes to the
voicemail system. This is bad because if caller dont leave a message
asterisk will send a SMS message.
How can I check if the caller leave the message and the execute the script?
Thanks
Bart
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RE: [Asterisk-Users] Best Linux for Asterisk
Peter Corlett [EMAIL PROTECTED] wrote: Eric Kirkland [EMAIL PROTECTED] wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? Best is highly subjective, and asking is likely to provoke a holy war ;) Agreed. Having said that, Gentoo is clearly the best. :-) So many people mention Gentto, however I have found very little help/guide for this particular distro. Any pointers ? Umar ___ 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] hang up when going to voicemail
Very welcome, Glad to have helped. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Simpson Sent: 25 July 2004 17:46 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] hang up when going to voicemail Doh! The reason it changed when I upgraded is because I was compiling VM with Mysql, and I had the mailbox definitions in the voicemail.conf flat-file. I put the definition in the SQL database and it works fine, now. :-/ thanks for kicking me into the right direction :) yours, matthew Are you sure you have a mailbox for this number ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Simpson Sent: 23 July 2004 16:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] hang up when going to voicemail I have a little menu set up where hitting 1, 2, or 3 places the call through to a cellular phone over IAX. That works. However, if caller hits 4 to go into voicemail, the system hangs up. Am I doing something wrong in the dial plan, or is this a CVS change? I had no trouble with this until I upgraded to about 07/21 CVS, and I'm on 07/23 [latest] now with same results. My dial plan: [txlink] exten = s,1,Answer exten = s,2,Background(/txlink/txlink-main) exten = 1,1,Dial(IAX2/:[EMAIL PROTECTED]/12149490280) exten = 1,2,Hangup exten = 2,1,Dial(IAX2/:[EMAIL PROTECTED]/14693373687) exten = 2,2,Hangup exten = 3,1,Dial(IAX2/:[EMAIL PROTECTED]/18174017579) exten = 3,2,Hangup exten = 4,1,VoiceMail(s2147649296) exten = 4,2,Hangup exten = t,1,Goto(txlink,s,2) exten = i,1,Playback(invalid) [didin] exten = 2147649296,1,Dial(SIP/2147649296,15) exten = 2147649296,2,Goto(txlink,s,1) exten = 2147649296,3,Hangup Here is console output: -- Executing Goto(SIP/2147649296-fb41, txlink|s|1) in new stack -- Goto (txlink,s,1) -- Executing Answer(SIP/2147649296-fb41, ) in new stack -- Executing BackGround(SIP/2147649296-fb41, /txlink/txlink-main) in new stack -- Playing '/txlink/txlink-main' (language 'en') == CDR updated on SIP/2147649296-fb41 -- Executing VoiceMail(SIP/2147649296-fb41, s2147649296) in new stack -- Executing Hangup(SIP/2147649296-fb41, ) in new stack == Spawn extension (txlink, 4, 2) exited non-zero on 'SIP/2147649296-fb41' ___ 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] hang up when going to voicemail
Are you sure you have a mailbox for this number ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Simpson Sent: 23 July 2004 16:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] hang up when going to voicemail I have a little menu set up where hitting 1, 2, or 3 places the call through to a cellular phone over IAX. That works. However, if caller hits 4 to go into voicemail, the system hangs up. Am I doing something wrong in the dial plan, or is this a CVS change? I had no trouble with this until I upgraded to about 07/21 CVS, and I'm on 07/23 [latest] now with same results. My dial plan: [txlink] exten = s,1,Answer exten = s,2,Background(/txlink/txlink-main) exten = 1,1,Dial(IAX2/:[EMAIL PROTECTED]/12149490280) exten = 1,2,Hangup exten = 2,1,Dial(IAX2/:[EMAIL PROTECTED]/14693373687) exten = 2,2,Hangup exten = 3,1,Dial(IAX2/:[EMAIL PROTECTED]/18174017579) exten = 3,2,Hangup exten = 4,1,VoiceMail(s2147649296) exten = 4,2,Hangup exten = t,1,Goto(txlink,s,2) exten = i,1,Playback(invalid) [didin] exten = 2147649296,1,Dial(SIP/2147649296,15) exten = 2147649296,2,Goto(txlink,s,1) exten = 2147649296,3,Hangup Here is console output: -- Executing Goto(SIP/2147649296-fb41, txlink|s|1) in new stack -- Goto (txlink,s,1) -- Executing Answer(SIP/2147649296-fb41, ) in new stack -- Executing BackGround(SIP/2147649296-fb41, /txlink/txlink-main) in new stack -- Playing '/txlink/txlink-main' (language 'en') == CDR updated on SIP/2147649296-fb41 -- Executing VoiceMail(SIP/2147649296-fb41, s2147649296) in new stack -- Executing Hangup(SIP/2147649296-fb41, ) in new stack == Spawn extension (txlink, 4, 2) exited non-zero on 'SIP/2147649296-fb41' ___ 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: MYSQL_FRIENDS and IAX problem
It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 07:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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] Re: Re: MYSQL_FRIENDS and IAX problem
Bug report might be a good idea, I just dropped the issue as I could do without using IAX. I am sure others may not have that flexibility. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 19:10 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem hmm - this is the bad thing about open source etc. Should we make a bugreport ? or are we just doing something wrong ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- usedcanon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 07:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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 ___ 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] Important note for AGI with PHP newbies
Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Alpert Sent: 15 July 2004 22:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Important note for AGI with PHP newbies I say this note is important only because I (a AGI PHP newbie) was tormented by this problem for many an hour, even though I'm sure it's documented somewhere or obvious to more experienced users. So as I was experimenting with AGI in PHP scripting I was baffled by why Asterisk was properly receiving AGI commands written to stdout but always returning 510 invalid command to the stdin. For example, doing a verbose command would properly show the message in the asterisk CLI but it would return 510 invalid. Also, commands like GET DATA and GET DIGIT would not work at all, even though the CLI would report Playing file my-file. After spending literally hours fumbling around with different buffer flushing schemes in PHP to no avail (I already knew I was doing the flushing properly anyway) and sending an email to this email list, I finally wrote a script that actually returned the proper values in stdin. The thing that fixed it was putting the -q argument in the #!/usr/local/php -q thing that you put on the first line of your script file. Apparently this argument tells PHP to supress HTML headers or something according to the documentation. I'm still not %100 sure why this would make a difference, but it does! Everybody is probably already aware of this, but I wasn't, so I hope this helps any other newbie who had the same problem. -nate alpert I wish more people did the same. I mean share there positive experience, ideally it would be good if you can add your experience to the wiki (that's where I try to do the same). Most people (IMHO) ask questions and don't bother sharing their resolutions with others. Umar ___ 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] Meetmee feature - Possible.
Hi Everybody, Is it possible to create a meetme conference by dialing out to an extension. I have a scenario where I want it to be possible for someone to listen into conversation to a particular extension. What I am thinking of is the possibility of an incomming call creating a conference. So the caller automatically joins a conference and the person called gets a ring on there phone and when they answer they effectively join a two party conference. Now the manager can join the conference in listen mode to listen into this call. I guess what I am asking for is a simmillar feature as Zapbarge but for SIP. Thanks in advance. ___ 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] using asterisk voicemail with a class 5 softswitch
Sylantro. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Carr Sent: 12 July 2004 14:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch Hi, which IP Centrex setup are you using? Gary I am using asterisk as a voicemail server for our IP Centrex SoftPBX. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 22:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch when you say you have integration what exactly do you mean? are you using asterisk as the voicemail system for a class 5 switch? On Friday 09 July 2004 15:45, usedcanon wrote: I have integration. Asterisk is upto the task however you may need to do some work arounds. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 20:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch anyone have any idea on the compatibility of asterisk voicemail with a class 5 switch that can do SIP (in particular the MetaSwitch VP3500)? -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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 -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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] VoiceMail - Catching caller hangup
When the caller hangsup, asterisk will jump to extension h if exists.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Frank
Sent: 12 July 2004 20:56
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VoiceMail - Catching caller hangup
I am trying to do some cleanup after a voice mail is left. But if the
caller just hangs up (user does not press # to send/complete the message
or wait for timeout to be reached), the voicemail seems to do an
immediate hangup and does not step through the rest of the context.
In the case below, priority 3 never gets executed if the caller just
hangs up after speaking their voice message.
exten = s,1,ResponseTimeout(30)
exten = s,2,VoiceMail(${ARG1}${ARG2})
exten = s,3,GoToIf($[${ARG3} = 0]?s|5)
how can I trap this condition so that I can still get priority 3
executed after voicemail exits?
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RE: [Asterisk-Users] call Intrude
search in the wiki for zapbarge. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Boardman Sent: 12 July 2004 20:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] call Intrude Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb ___ 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] IVR Menu and VoiceMail quality
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent: 11 July 2004 08:35 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality Yep I sure did, damn upstream pipe gets so congested I had to drop it to about 75% to keep from dropping packets... Seems to be working excellently, I tried downloading a large file and doing some interactive SSH with no no noticeable degradation... I'd say we have a winner. Installing and running Ztdummy seems to have done the trick, I cannot tell a difference between the quality over VoIP and POTS now, it's excellent... So for anyone confused on this issue, if you run a pure VoIP setup with no digium hardware and you want asterisk to do ANYTHING, not just MOH and MeetME you MUST have some kind of timing source, either ZTDummy or ZapRTC installed. Especially for doing VoiceMail, that seemed to be the worst for some reason... This was very confusing for me because the wiki says that it's only for MOH and MeetME, that's simply not true or at least not in my experience. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 10, 2004 6:21 AM Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality On 9 Jul 2004 at 14:08, Chris Shaw wrote: Thx Jay, I hope this is not a too FAQ... I really did try to look it up first but I saw s many conflicting things about timing... one person says no you absolutely do not need ztdummy or a digium card to make IVR/Voicemail work, others say you need it for everything... I tend to believe the latter since it seems to be more of a timing issue than a bandwidth issue... What I can't figure out though is if it's a timing thing, shouldn't calls on my local net be crappy too? When I log into voicemail from my ip phone it's perfect... when I call home from out of town it sounds like crap unless I play with the nice values or restart asterisk... Just a thought, when setting up your QOS, did you make sure that the maximum usage was slightly below your actual pipe size? Matt - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 1:48 PM Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality AFAIK, it's needed anytime asterisk streams audio... Which is meetme, MOH and of course voicemail and IVR. My Asterisk system had lousy IVR quality until I plugged in the FXO card and loaded Zaptel. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Friday, July 09, 2004 3:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality I thought it was only needed for MeetMe and MOH? - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 12:21 PM Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality Do you have ztdummy loaded? -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Friday, July 09, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IVR Menu and VoiceMail quality I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP - PSTN and PSTN - SIP calls, however when I or anyone else calls from PSTN - * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell that it's actually VoIP. Sometimes it's so bad that I can't understand what Allison's saying at all... Calls on the same network sound just fine... I know what you're thinking, it's a congested link, and that may be but I've noticed that if I play with the nice value of asterisk, it seems to help. Setting nice to 0 seems to work the best, I tried -20 and it was the worst... I have implemented QoS on my network and have given any and all asterisk packets priority. As I said actual calls are crystal clear so I believe it to be a problem with Asterisk itself or the machine it's running on. Possibly some bottleneck somewhere. I realize that since it's going over the public internet, the occasional dropped packet is to be expected, but what's frusterating is that I can get crystal clear menus sometimes even when my network is fully loaded and other times when it's perfectly quiet it sounds absolutely horrible... Here are the machine's specs if that helps: AMD Athlon 1Ghz (Old Thunderbird core) Asus A7V600 128MB DDR-266 RAM 450GB storage (4 IDE drives in an LVM array) *grin* Pure VoIP, no digium hardware Internet connection is
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I was going to keep out of this (was interesting to read, as I have dealt
(Bwith simmillar situation) however I would like to add just this one commnet.
(B
(BTry to better understand asterisk than to throw about your money. What you
(Bwant to do is perfectly possible with asterisk there is no need to add a new
(Bconfusing feature.
(B
(BAs for your bounty, donate it to the wiki ! :-)
(B
(BUmar.
(B
(B-Original Message-
(BFrom: [EMAIL PROTECTED]
(B[mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan
(BSent: 11 July 2004 09:51
(BTo: [EMAIL PROTECTED]
(BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(BI accept your views.
(B
(BI have a specific requirements, can you help to attain the same.
(BIn our business we have 25 employees handling customer service.
(B
(BI want to add or remove employees in the customer service so does the
(Bdevices connected to it.
(BI don't want to make any changes in the asterisk, and all I need is to plug
(Bin the VoIP Phone and start handling the customer service. I would like to
(Bdo for as many employees as I want without any problems.
(B
(BCan you think of a better solution?
(B
(B-Kannaiyan.
(B
(B- Original Message -
(BFrom: "Sunrise Ltd" [EMAIL PROTECTED]
(BTo: [EMAIL PROTECTED]
(BSent: Sunday, July 11, 2004 9:15 AM
(BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(B When I call a SIP user, the phone should ring in more
(B than one
(B extentions. Also more than one phone should be able to
(B register with
(B asterisk. Right now it is not the case.
(B
(B There is no issue here. You seem to be confused, that's
(B all.
(B
(B A SIP account is a SIP account and an extension is an
(B extension. You can assign an extension to an account (or
(B to multiple accounts) and the tool for that is the dial
(B command.
(B
(B However, there is no implicit assignment between an
(B extension and an account and that is good so. This should
(B not be changed because it would harm Asterisk's
(B flexibility and manageability.
(B
(B
(B This type of situations might be needed in call centres.
(B
(B Called 12345
(B |---(12345) Ringing
(B |---(12345) Ringing
(B |---(12345) Ringing
(B
(B As I said, you are confusing extensions with accounts. The
(B first "12345" is an extension, the three "(12345)"s are
(B accounts. Those are different layers, don't mix them up.
(B
(B You should always be able to distinguish between devices,
(B even if they are assigned the same phone number. In fact,
(B in a call centre you'd be using a call queue. It would be
(B rather nonsensical for a call queue management to have to
(B distinguish between multiple identical agents.
(B
(B Therefore, setting up multiple devices with the same
(B account credentials is not a good idea, especially not in
(B a call centre. Each device and each agent should have
(B their own unique account credentials and assigning
(B extensions to them should always be done through the
(B dialplan and only the dialplan.
(B
(B Asterisk has been designed this way. It is a good design.
(B It should NOT be changed nor undermined.
(B
(B You may want to do something like this ...
(B
(B [GLOBALS]
(B
(B A-GROUP = SIP/2001 SIP2002 SIP/2003
(B
(B B-BROUP = SIP/jdoe SIP/dflint SIP/bsmith
(B
(B ...
(B
(B
(B [Support]
(B
(B exten = 12345,1,Dial(${A-GROUP},30,r)
(B ...
(B
(B exten = 54321,1,Dial(${B-GROUP},30,r)
(B ...
(B
(B
(B There is of course an issue when you want to let different
(B phones start ringing at different times, for example, the
(B first phone is supposed to start ringing immediately and
(B the other two are only to join in if the first phone
(B hasn't been picked up in 10 seconds, like so
(B
(B exten = 12345,1,Dial(${JDOE},10,r)
(B exten = 12345,2,Dial(${JDOE}{DFLINT}${BSMITH},20,r)
(B
(B This works but if JDOE was to pick up right between those
(B two dial commands, then it will have been too late for the
(B first and JDOE will be "on the phone" for the second dial
(B command, so there is some room for improvement. A bounty
(B might better be spent on solving this little problem.
(B
(B Also, Asterisk supports call groups and pickup groups.
(B Indeed, there have been some bugs with those features and
(B I am not sure if they have have been fixed, but if they
(B haven't, then it would again make more sense to put the
(B bounty on fixing those rather than creating an ugly
(B workaround.
(B
(B
(B I feel this is a great feature
(B
(B I don't and if you spent some more time with Asterisk and
(B immerse its philosophy, then you'll very likely change
(B your mind.
(B
(B in other SIP proxy server this can be done easily
(B
(B Asterisk is not a SIP proxy. It's a telephone exchange.
(B
(B i mean its default 1 or more phone could be registered
(B at 1 number (12345) and resulting same effect
[Asterisk-Users] Hardware for sale / donate
I have a lot of PC bits and pieces that I no longer need. I am moving to new premisses soon so I thought I offer them up for sale. Usually I would sell stuff through eBay but I dont really have much time for that right now. Anyway what I would like to do is offer it here and donate ½ of the proceeds to the wiki. To give an idea of what I have here is a quick lists ... 3com 10/100 ethernet cards - lots ! ATI, PCI and AGI cards - a few PII/PIII SBC (Single board computer) - about 4 with matching backplanes PII/PIII - two each PII 400 and PIII 600e 4 GB SCSI Hard disks - about 6 CD ROM, Floppy drives - about 10 or so. This is a list from memory, if there is any interest I can make a more complete list. Umar. ___ 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] using asterisk voicemail with a class 5 softswitch
I am using asterisk as a voicemail server for our IP Centrex SoftPBX. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 22:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch when you say you have integration what exactly do you mean? are you using asterisk as the voicemail system for a class 5 switch? On Friday 09 July 2004 15:45, usedcanon wrote: I have integration. Asterisk is upto the task however you may need to do some work arounds. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 20:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch anyone have any idea on the compatibility of asterisk voicemail with a class 5 switch that can do SIP (in particular the MetaSwitch VP3500)? -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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 -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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] using asterisk voicemail with a class 5 softswitch
I have integration. Asterisk is upto the task however you may need to do some work arounds. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 20:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch anyone have any idea on the compatibility of asterisk voicemail with a class 5 switch that can do SIP (in particular the MetaSwitch VP3500)? -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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] xlite calls not approved
It would be nice if you were to let others know how you fixed the problem in case they are having same issues now or in future. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of CHS Sent: 09 July 2004 21:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] xlite calls not approved ok, I've finally got it working. I can get to the demo extension '1000' and I hear the voice, etc.. only one problem, I can't seem to hit any of the demo extensions (like 2 for more detailed info, etc..) It would be nice if you were to let others know how you fixed the problem in case they are having same issues now or in future. ___ 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
checkout .. http://www.automated.it/asterisk/ and http://knopsterisk.com/ Feedback back to the forum once you make progress would be useful. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 08 July 2004 03:50 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Small Linux Distro Does anyone have a current, stripped linux distro which has only asterisk and net drivers? If so do you have it available somewhere? I guess also, my question could be, does anyone know of a small distro, which will run asterisk. When I say small I mean 700Mb Also, anyone got any sites on hand which would point to ways to make linux start up faster? (BTW this is all in aid of making Asterisk boxes, with LCDs and buttons as opposed to keyboard and screen - i will also write an interface for Asterisk to LCDproc, so that it can be controlled from buttons mounted next to the screen, and make it GPL). Any help, pointers greatly appreciated. Matt ___ 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] FINALLY! a good book about Asterisk.
-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. ___ 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] Slackware 10.0 and asterisk and 2.4 vs 2.6
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Zdenek Bouresh Sent: 08 July 2004 22:33 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6 mattf wrote: Hello, I am running Asterisk on Slackware 10.0 with the 2.4 kernel(default kernel) and it is very happy. Don't see too much difference from 9.1 except for the fact that most of the binutils have been updated and several of them run differently now(top, ps, ...) Haven't tried the 2.6 kernel yet, but may try it later. MATT--- -Original Message- From: Joe Baptista [mailto:[EMAIL PROTECTED] Sent: Thursday, July 08, 2004 10:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6 I'm installing the new Slackware 10.0 distribution - but not sure if i should go with the 2.4 kernal - which i think is the default install - or the new 2.6 kernal? anyone running * and slackware 10.0 with 2.6 kernal? thanks joe ___ 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 Slackware 10 and kernel 2.6.6 and 2.6.7 run fine . What about MySQL support for SIP friends, CDR and Voicemail ? do they work. Umar. ___ 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] using SetCDRUserField in an AGI script
Took me some time to get around to check this.
Anyway for the benifit of everyone else. It worked after implementing your
suggesstion. Thanks for your help.
Umar.
p.s I will update the wiki with this information.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Leo Ann Boon
Sent: 17 June 2004 02:13
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] using SetCDRUserField in an AGI script
Did you set
userfield=1
in cdr_mysql.conf?
Umar Sear wrote:
Hi I am trying to use SetCDRUserField in an agi script
but with no success.
I am using the CDR mysql addon, however I can't see it
being at fault as my attempt is not doing anything to
the CVS CD either.
has anyone used this, any hints guidence would be
greatly appreciated.
The syntax I am using is like so ..
res=DoExec('SetCDRUserField','12345');
and then dialing the relevant extention.
Thanks
Umar.
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[Asterisk-Users] How to use return value in extensions.conf
Hi,
I am trying to implement a dialplan in which the user is notified of a
missed call, if no voicemail is left.
Basically what I would like to achieve is something like ...
exten = _0207XXX,1,DIAL(SIP/${EXTEN},15)
exten = _0207XXX,2,HasNewVoicemail(${EXTEN:[EMAIL PROTECTED]:INBOX|msgcount)
exten = _0207XXX,3,Voicemail(u${EXTEN:4})
exten = _0207XXX,4,HasNewVoicemail(${EXTEN:[EMAIL PROTECTED]:INBOX|msgcount2)
exten = _0207XXX,5,GotoIf($[${msgcount2}${msgcount1}]?7:6)
exten = _0207XXX,6,Send an email or something. !
exten = _0207XXX,7,Hangup
However when the user hangsup the rest of the dial plan seems to be skipped.
Any ideas ? suggestions.
Umar.
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RE: [Asterisk-Users] How to use return value in extensions.conf
I am happy using AGI, however the dialplan does not seem to work. What
should I expect the priority to jump to when the caller hangsup during
voicemail greeting playback.
Thanks
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chad Scott
Sent: 04 July 2004 18:42
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to use return value in extensions.conf
This looks like a job for AGI...
I'd do something like
exten = _0207XXX,1,Dial(SIP/$EXTEN},15)
exten = _0207XXX,2AGI('missed-call-email.agi')
exten = _0207XXX,3,Voicemail(u${EXTEN:4})
exten = _0207XXX,4,Hangup
exten = _0207XXX,102,AGI('missed-call-email.agi')
...etc...
On Jul 4, 2004, at 5:03 AM, usedcanon wrote:
Hi,
I am trying to implement a dialplan in which the user is notified of a
missed call, if no voicemail is left.
Basically what I would like to achieve is something like ...
exten = _0207XXX,1,DIAL(SIP/${EXTEN},15)
exten =
_0207XXX,2,HasNewVoicemail(${EXTEN:[EMAIL PROTECTED]:INBOX|msgcount)
exten = _0207XXX,3,Voicemail(u${EXTEN:4})
exten =
_0207XXX,4,HasNewVoicemail(${EXTEN:[EMAIL PROTECTED]:INBOX|msgcount2)
exten = _0207XXX,5,GotoIf($[${msgcount2}${msgcount1}]?7:6)
exten = _0207XXX,6,Send an email or something. !
exten = _0207XXX,7,Hangup
However when the user hangsup the rest of the dial plan seems to be
skipped.
Any ideas ? suggestions.
Umar.
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RE: [Asterisk-Users] How to use return value in extensions.conf
I was not thinking straigth I guess.
the behaviour is default, when the caller hangs up, the dial plan jumps to
exten h if there is one.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent: 04 July 2004 22:17
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] How to use return value in extensions.conf
I am happy using AGI, however the dialplan does not seem to work. What
should I expect the priority to jump to when the caller hangsup during
voicemail greeting playback.
Thanks
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chad Scott
Sent: 04 July 2004 18:42
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to use return value in extensions.conf
This looks like a job for AGI...
I'd do something like
exten = _0207XXX,1,Dial(SIP/$EXTEN},15)
exten = _0207XXX,2AGI('missed-call-email.agi')
exten = _0207XXX,3,Voicemail(u${EXTEN:4})
exten = _0207XXX,4,Hangup
exten = _0207XXX,102,AGI('missed-call-email.agi')
...etc...
On Jul 4, 2004, at 5:03 AM, usedcanon wrote:
Hi,
I am trying to implement a dialplan in which the user is notified of a
missed call, if no voicemail is left.
Basically what I would like to achieve is something like ...
exten = _0207XXX,1,DIAL(SIP/${EXTEN},15)
exten =
_0207XXX,2,HasNewVoicemail(${EXTEN:[EMAIL PROTECTED]:INBOX|msgcount)
exten = _0207XXX,3,Voicemail(u${EXTEN:4})
exten =
_0207XXX,4,HasNewVoicemail(${EXTEN:[EMAIL PROTECTED]:INBOX|msgcount2)
exten = _0207XXX,5,GotoIf($[${msgcount2}${msgcount1}]?7:6)
exten = _0207XXX,6,Send an email or something. !
exten = _0207XXX,7,Hangup
However when the user hangsup the rest of the dial plan seems to be
skipped.
Any ideas ? suggestions.
Umar.
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RE: [Asterisk-Users] looking for newbie resources
www.voip-info.org ! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven M. Sawczyn Sent: 04 July 2004 19:53 To: Asterisk-Users Subject: [Asterisk-Users] looking for newbie resources 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. 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. Thanks in advance, Steve ___ 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] Detect more than one type of DTMF for calls to voicemail
Hi, I was thinking of raising a feature request, but thought I ask here first. Basically I am looking for options to allow more than one type on incomming dtmf types for incomming voice calls. I am using purely SIP on my system and to the best of my knowledge there is only option to specify one type of DTMF in SIP.conf Feedback, suggestions will be appreciated. Umar ___ 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] using asterisk as sip registrar is not working for me
Have you tried pinging the phone from the asterisk box to see if there is connectivity ? if so try doing a network capture to see if asterisk is recieving registeration packets at all ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of smadi Sent: 19 June 2004 00:53 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] using asterisk as sip registrar is not working for me hi; i have the following topolgy: asterisk box set with public ip address 1.2.3.4 i have a snom200 sip phone that resides on a subnet with 192.168.0.10 address which i know have worked previously with vocal now my sip.conf file looks as follows: == [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls [phone1] type=friend host=dynamic dtmfmode=inband mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Me 2124 = the phone does not seem to register. I tried to include the 192.168.0.10 defaultip address and i tried putting the static ip address of the dhcp server which is the gateway to which is the snow phone connected. is there anything specific that i must run to get the phone working? thanks m. smadi zia.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] how can I catch
How to catch some incoming call Date: Tue, 15 Jun 2004 19:37:28 +0100 Message-ID: [EMAIL PROTECTED] MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2911.0) X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2800.1409 In-Reply-To: [EMAIL PROTECTED] Importance: Normal A simple search would probably yeild hundreds of examples. This is a pretty well documented situation. Search on www.voip-info.org Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miroslav Nachev Sent: 15 June 2004 11:38 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] how can I catch How to catch some incoming call Hello, I have a question about the configuration of the SIP telephone. The situation is following: We have two SIP telephones. One of them is configured to answer the incoming calls from FXO or other directions. If there is no one available to answer to the ringing phone, I would like to take (redirect) the call and answer to it using my SIP telephone. Could you give me any ideas how to set up this configuration - how can I catch the incoming call. Thank you in advance. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ 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 V. SER
I think I do agree with your assessment that *BSD are more stable that linux, no disrespect meant to linux as I think it is wonderful in its own right. What version of FreeBSD ? BSD you are using ? I am looking to build an athlon 64 server soon and am wondering if FreeBSD would be a better option. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Aaron J. Angel Sent: 15 June 2004 20:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ASTERISK V. SER Usedcanon wrote: Just out of interest (as I am a freeBSD fan) why more stable on BSD ? I have no idea, it just seems to run better on *BSD. I'm still trying to investigate that myself. Perhaps I'm just inept when it comes to Linux, but it has never run decently for me -- I've always had problems with whichever distro I try. This time it seems to be the network card mostly, but then I get similar response from the console every now and then, so maybe it's not the NIC. Maybe I should have rephrased that or left it out, as it's probably not Asterisk that is more stable, technically speaking. ___ 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 V. SER
Just out of interest (as I am a freeBSD fan) why more stable on BSD ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Aaron J. Angel Sent: 14 June 2004 15:24 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ASTERISK V. SER Joshua Colp wrote: I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution Linux PBX solution is such a narrow point of view. Asterisk also runs on *BSD; yes, conferencing and MP3s are a bit sketchy due to lack of timer (for now), but the rest of it is fully functional (and more stable) on *BSD. ___ 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] Polycom IP 600
I have a polycom IP600 for testing and will try it out if possible. Not sure at the moment if the phone as MGCP or SIP software on it. Btw, has polycom released the IP600, I was under the impression that it was still in testing. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Mandel Sent: 14 June 2004 17:24 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom IP 600 I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if anyone can confirm that this works with the polycoms. I know the 7960s support this, but I want to make sure the Polycom sales team wasn't just saying Yes to make the sale. Any comments are appreciated. -Eric -Original Message- Subject: fwd on busy when calling multiple extensions at once Chris A. Icide wrote: IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I base this off of having had both an IP600 and a 7960. The two advantages the 7960 had over the IP600 was appearance and ease of configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down. Now, you MAY want to try registering all 6 lines on the polycom to the same line and see if the phone handles that as well as the cisco. If it does, then you are set. Otherwise, you will need some complex configuration work in your extensions.conf to achieve what you are looking to achieve. Some thoughts: What do you want to happen when one of the call takers has all 6 lines in use? Have you considered using queues to do what you need? -Chris On 10:08 AM 5/22/2004, Brian Cuthie wrote: You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ 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] prepaid running mysql
What exactly do you want it doing, I am working on a prepaid AGI script. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of HCQ Sent: 14 June 2004 15:33 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] prepaid running mysql any prepaid app running in MYSQL? I already have mysql and dont want to add postgres.. thanks anybody. H.C ___ 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] Sayson IP Phones?
You are right, I am using (or hoping too!) as an MGCP device. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin P. Fleming Sent: 14 June 2004 17:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sayson IP Phones? Michael Graves wrote: No, sir. Have not seen IP300. However, a friend loaned me a IP600 for evaluation. I have yet to figure out its support and configuration. Looks like a nice instrument. The Polycom SoundPoint IP300 is not an SIP-capable phone; H.323 and MGCP only. ___ 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] SIP prepaid
I am doing something for someone. I counld not find anything that met the spec. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of HCQ Sent: 14 June 2004 20:28 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP prepaid I want to have sip prepaid (not calling card) users, is there anything already developed for that matter? Thanks . HC ___ 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] Wiki now based on CVS head
Would it be a good idea for some on to add skeleton pages for new featrues so that atleast more people know about the new featrues and fill in the pages once they have had a chance to check/test. Just a thought. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: 13 June 2004 22:00 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] Wiki now based on CVS head Due to the dismissal of the stable-1.0 cvs source code, I've changed policy of the Asterisk Wiki - we now document CVS head. I would like all contributors to document which version of Asterisk (date) an addition was applied to, so readers can find out if a new function works with their version or not. There's a lot of missing documentation of new functions and options in CVS head up there, so please help us update. Next time we fork, we'll consider changing this policy. Thank you for your help with keeping this valuable community resource up-to-date. /O ___ 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] Sayson IP Phones?
Has anyone had a chance to test the new polycom IP300 ? I think it is not a release product yet. I just had one land on my desk this Friday and will be testing it soon. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: 13 June 2004 22:21 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sayson IP Phones? At 2:05 PM -0400 on 6/13/04, Michael Graves wrote: Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] [snip] No, they haven't started to ship yet. Supposedly they will ship in some quantity next month, according to their sales and marketing people, who I've been in contact with on this and other topics. Even when they do ship, they're only coming with MGCP and H.323, and SIP isn't anticipated to be shipment-ready until August (read: September.) Other options in the interim: Polycom 600, Zultys (as you mentioned) and of course Cisco. I'm really looking forward to possible srtp in * for the Zultys and Sipura if that can be tested and qualified... JT ___ 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] ssh key problem
Try logging on to the ftp server on the machine itself. It could be a permissions issue. Umar. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean CollinsSent: 12 June 2004 01:54To: [EMAIL PROTECTED]Subject: [Asterisk-Users] ssh key problem Hi Ive need to reinstall my asterisk software (hard drive failure). Im back and running to a make samples state. I have backed up all of my conf files (ok so they were about a week old but much better than starting from scratch), the problem I am having is with WS_FTP Pro. Basically I used to connect to my asterisk server using this software no problems just using root as username and password but I can no longer connect to the new installation. I also use Putty to connect from the same windows machine and no problems with using this. For some reason WS_FTP Pro will not all me to connect with new install, I have deleted WS and reinstalled twice but still no luck. I think it may have something to do with SSH keys. Any thoughts? Cheers, Dean
RE: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
Quite simple really,
You could do the following assuming your area code is 0207 (london !)
exten = 9NXXNXXX,1, Dial(SIP/0207${EXTEN})
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter
Sent: 12 June 2004 09:15
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code
for 7 digit dialing
How do you prepend. I want to be able to dial 7 digits instead of of
11 for local calls.
Can someone post there extensions.conf part that is relavent?
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RE: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
I know :-), it was just an example.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Lee
Sent: 12 June 2004 11:06
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area
code for 7 digit dialing
usedcanon wrote:
Quite simple really,
You could do the following assuming your area code is 0207 (london !)
exten = 9NXXNXXX,1, Dial(SIP/0207${EXTEN})
Umar.
The London code is 020 the 7 or the 8 is part of the local number now.
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RE: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
looks fine to me
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter
Sent: 12 June 2004 12:48
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area
code for 7 digit dialing
Does this look right
exten = _9NXX,1,SetCallerID(831-XXX-)
exten = _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten = _9NXX,3,Congestion
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RE: [Asterisk-Users] MWI on Cisco ATA-186 (SIP)
It should go in sip.conf the context is whatever context you specified in voicemail.conf Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] Cisco ATA-186 Firmware upgrade
There probably are a number of fixes. I have not used the ATA's for some time, however as the saying goes .. If it ain't broke don't fix it. So if it is working for you don't bother. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA-186 Firmware upgrade I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ 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] Broadvoice and DTMF
Depends how you have it configured. To the best of my knowledge asterisk supports, inband, info and RFC2833 Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: 12 June 2004 14:26 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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] MWI on Cisco ATA-186 (SIP)
If you have more than one context than yes, otherwise I believe it will work with out it. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:45 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) the mail=1234 seems to have worked... is it necessary to do the [EMAIL PROTECTED] I dont think i set a contect (as there is only 2 mailboxes) so would it be default.. On Jun 12, 2004, at 6:40 AM, usedcanon wrote: It should go in sip.conf the context is whatever context you specified in voicemail.conf Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] FW: question about prepaid app_prepaid
Hi, I was just going to say, the latest dial app has a lot more options, some very nice features. As for locking the account whilst in use, I do the same using a database and preventing further calls from the same account. I am actually using Pascal, to write AGI script, my requirement were simpler than whats been addressed in app_prepaid. Basically I am not usiung calling cards, rather callerid as the account. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: 11 June 2004 06:09 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Hi, I just looked at the CVS. It has some cool stuff in it. A lot of changes! I have looked at the ast_channel_bridge() function to see if it is any different but will take me a while to hash out. But it looks like it would be easy to implement a real time multi call credit system in it. For now my patch is really just for someone who wanted to use the Stable 1.0 Branch - rather than a development version. I personally like to stick with the latest stable release and base my code around it. Then usually I can just blame *my* code if something goes wrong. I'm going to snoop more into the CVS now. ^_^ S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: Thursday, June 10, 2004 9:45 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid I haven't looked at the CVS source yet - I will take a look and see if it's the similar or different. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Murphy Sent: Thursday, June 10, 2004 9:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Just one question about the B() option: When you say that it limits a call to X seconds, from the time the call is bridged, as opposed to from the time the call is dialed, is that comparing it to the L() option? I haven't plumbed the depths of the L() command in the current CVS source, but is this the difference you are referring to? murf [From the apps/app_dial.c source code: 'L(x[:y][:z])' -- Limit the call to 'x' ms warning when 'y' ms are left (repeated every 'z' ms)\n -- Only 'x' is required, 'y' and 'z' are optional.\n -- The following special variables are optional:\n ** LIMIT_PLAYAUDIO_CALLER(default yes) Play sounds to the caller.\n ** LIMIT_PLAYAUDIO_CALLEEPlay sounds to the callee.\n ** LIMIT_TIMEOUT_FILEFile to play when time is up.\n ** LIMIT_CONNECT_FILEFile to play when call begins.\n ** LIMIT_WARNING_FILEFile to play as warning if 'y' is defined.\n -- 'timeleft' is a special sound macro to auto-say the time left and is the default.\n\n ___ 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] FW: question about prepaid app_prepaid
I believe it is from when its bridged through. Basically how long it takes the far end to answer is ignored. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: 11 June 2004 10:29 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Hi, Don't try to use my patch with the latest app_dial. It will only work with Release 1.0. Mine is just very clean and simple implementation to force a disconnect X seconds after a call was bridged. I was skimming though the latest source tonight and you are right, lots of nice features. I don't know if the L() option starts counting from when the call was made or after it was bridged though. Does anyone know? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of usedcanon Sent: Friday, June 11, 2004 1:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Hi, I was just going to say, the latest dial app has a lot more options, some very nice features. As for locking the account whilst in use, I do the same using a database and preventing further calls from the same account. I am actually using Pascal, to write AGI script, my requirement were simpler than whats been addressed in app_prepaid. Basically I am not usiung calling cards, rather callerid as the account. Umar. ___ 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] Background Playback fails
have
tried specifying the full path ?
Umar
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Tim
GuySent: 11 June 2004 12:39To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Background
Playback fails
Hi
Guys.
Ive had a lay off from Asterisk
for 12 months but I am starting to look into it again. I am not very Linux
savvy and found it hard going the last time. Ive started playing with it in
the last 3 weeks and I have to admit to making more head way this
time.
The first problem I'm stuck on and
I cant find a solution to is that sound files that I have recorded (be it by
creating a temp file from an extension number, or by recording in .wav) will
not play when I try to create a menu in the
extension.conf
The relevant lines
are:
exten =
s,1,Wait,2
exten =
s,2,Answer
exten =
s,3,Background(mainmenu)
The mainmenu.gsm is sitting in the
sounds folder.
I get an error on the console
saying:
-- Executing
Wait("SIP/timg-a4e6", "2") in new stack
-- Executing
Answer("SIP/timg-a4e6", "") in new stack
-- Executing BackGround("SIP/timg-a4e6", "mainmenu") in new
stack
Jun 11 12:33:00
WARNING[1209214400]: file.c:464 ast_openstream: File mainmenu
d
oes not exist
in any format
Jun 11 12:33:00
WARNING[1209214400]: file.c:752 ast_streamfile: Unable to
open
mainmenu (format ULAW): No such
file or directory
== Spawn extension (default, s, 3)
exited non-zero on 'SIP/timg-a4e6'
What am I doing wrong
guys???
Cheers
Tim
RE: [Asterisk-Users] Background Playback fails
Give it a go, path is usually not required, just wondering if it is looking in a different place. Also note that filename is case sensitive. Umar. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Tim GuySent: 11 June 2004 12:51To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Background Playback fails Not totally. I did read an archive that said DONT put .gsm or .wav on the end (I have to admit to trying) so I assumed paths was a no-no as well. Shall I try it?? Tim -Original Message-From: usedcanon [mailto:[EMAIL PROTECTED] Sent: 11 June 2004 12:40To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Background Playback fails have tried specifying the full path ? Umar
RE: [Asterisk-Users] cdr_addon_mysql.c
Are you sure you are following instruction from the wiki correctly .. ? http://www.voip-info.org/wiki-Asterisk+cdr+mysql I recently updated it after I had simmillar issues. I think the steps outlined are quite clear now. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ed Devine Sent: 11 June 2004 20:21 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cdr_addon_mysql.c Following the latest * CVS update, my MySQL was broken. Following the update, Asterisk-addons would compile fine, but when I ran asterisk I got the following error: ERROR[1202489024]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas I then tried using the patch (bug id 0001823) from bugs.digium.com, and found that Asterisk-addons would no longer compile, giving me the following errors: make clean ; make install rm -f *.so *.o .depend ../mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: parse error before '!' token cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:110: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:110: (Each undeclared identifier is reported only once cdr_addon_mysql.c:110: for each function it appears in.) cdr_addon_mysql.c:121: `mysql' undeclared (first use in this function) cdr_addon_mysql.c: In function `my_unload_module': cdr_addon_mysql.c:226: `mysql' undeclared (first use in this function) cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:380: `mysql' undeclared (first use in this function) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:422: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I then tried using and older version of cdr_addon_mysql.c, and it also would not compile, but gave me an entirely different set of errors: ]# make clean ; make install rm -f *.so *.o .depend ../mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [cdr_addon_mysql.o] Error 1 I'm stuck on an MySQL project until I can resolve this problem. I've even blown away my system (OS as well as asterisk) and reloaded everything fresh from CVS, and still no joy. Any suggestions would be greatly appreciated. ___ 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] 403 Forbidden between two softphones on same Asterisk
It would be useful if you could post your configs, otherwise it is impossible to know whats going on. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tor Houghton Sent: 02 June 2004 10:25 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 403 Forbidden between two softphones on same Asterisk Hi, I have two softphones connected to an Asterisk stable. I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a 403 Forbidden result, while a Diax97a on the same extension will just report Call disconnected by remote. The same is not true when 2000 calls extension 1000. Extension 1000 will ring, and is also able to pick up. Extension 2000 can also call external parties (routed through another Asterisk box), but again, external parties cannot call extension 2000 (they can call extension 1000, however!). I'm confident that I've made a mistake, but I just don't know where. Anyone have any ideas? Tor ___ 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] How to get the Called id with AGI
Hi
Angel,
I
assume you mean CALLERIDNUM (the number part of caller ID),
The
easiest thing in that case is to pass it as a parameter to your AGI script
extex = 500,1,AGI(myscript.py|${CALLERIDNUM})
in
your script you just used the argument passed as usual ( I am not a perl expert,
so not sure on the syntax there )
Hope
this helps.
Umar.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Angel
DiazSent: 10 June 2004 22:41To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] How to get
the Called id with AGI
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:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$callerid = $input{'callerid'};
$AGI-say_digits($callerid);
}
Thanks in advance,
Angel.
Do you Yahoo!?Friends. Fun. Try the all-new Yahoo!
Messenger
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I would be interested to share ideas, if you have guidence to offer I would be greatful Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: 11 June 2004 00:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Hi, I found that the PREPAID system didn't disconnect proper and tracked time from when you dialed not when your phone made connection. I ended up making my own system and had to modify the Dial app. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Umar Sear Sent: Thursday, June 10, 2004 9:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FW: question about prepaid app_prepaid Thanks to the lack of documentation, I decided to write my own AGI script (working but no where near complete) Look forward to replies and guidence on this topic. Umar. --- Yang Tao [EMAIL PROTECTED] wrote: Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.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 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] IAX Peers from MYSQL
Hi, I am having problems making calls from firefly when configured as IAX2 clients. Basically on the asterisk console I see a message saying, Rejected connect attempt from On firefly itself it gives an error, something like - no authority found... If I configure the accounts in iax.conf, calls work fine. The problem occurs when I am using the mysql databse. I know the database connectivity is working, as I can see the client register and the database records being updated with there IP address. Could I be making some basic mistakes. My setup at the moment is based on a recent CVS and my configs are based on the sample configs. Thanks Umar. ___ 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] IAX Peers from MYSQL
Hi, I am having problems making calls from firefly when configured as IAX2 clients. Basically on the asterisk console I see a message saying, Rejected connect attempt from On firefly itself it gives an error, something like - no authority found... If I configure the accounts in iax.conf, calls work fine. The problem occurs when I am using the mysql databse. I know the database connectivity is working, as I can see the client register and the database records being updated with there IP address. Could I be making some basic mistakes. My setup at the moment is based on a recent CVS and my configs are based on the sample configs. Thanks Umar. ___ 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] Zapata?
Clothes that you find hard to wear. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of hank smith Sent: 07 June 2004 04:11 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zapata? what is hadrware? - Original Message - From: Richard Neese [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 06, 2004 7:37 PM Subject: Re: [Asterisk-Users] Zapata? as for hadrware digitalnetworks has made a clone card . but only digium has made any majoor card changes. there have been 2 ne rev to the cards i kow of and you can rea d on the digium site about them. ___ 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] Dial plan help
Checkout http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial and http://www.voip-info.org/wiki-Asterisk+t+extension You could use extention t, which is reached after dial times out. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Simon Brown Sent: 07 June 2004 07:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dial plan help I wish to have outgoing calls try to use a SIP/IAX provider and if this fails, then fall back to PSTN and I am not sure how the dial plan should look. Can someone please post a sample of how it should look. Thanks in advance, Simon Brown ___ 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] Voicemail missing playback options
What version of software are you using. This is a recent feature. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Larry Hishon Sent: 07 June 2004 21:27 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail missing playback options According to my reading, Asterisk voicemail should have a playback option when I record my busy and unattended messages... it doesn't... Did I do something wrong... i.e. make an improper config selection, compile incorrectly, ...? ___ 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 moderator
I am having trouble finding links to continue after hangup based on dial.
Can you send me something on that ?
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bruce Marler
Sent: 05 June 2004 05:27
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Meetme with moderator
Florian,All,
So I did what was noted below, here what I run into though, how do I set the
DB entry for moderator after the moderator hangs up. I just read the other
posts to the list about continuing after a hangup, but that is based on the
dial command and not meetme.
Here is what I have setup now for my simple testing, basically i dial 100 it
asks for conf # , if they dial 101 it sets moderator code to 1, if they dial
102 it checks it and lets them in if it is 1, unfortuatnely it stays 1 even
after the moderator hangups:
[conferences]
exten = 101,1,Answer
exten = 101,2,Wait(1)
exten = 101,3,DBput(Moderator/5=1)
exten = 101,4,Meetme(5)
exten = 101,6,Hangup
exten = 102,1,Answer
exten = 102,2,Wait(1)
exten = 102,3,DBget(5Admin=Moderator/5)
exten = 102,4,Gotoif($[${5Admin} = 1]?5,1:5550001:1)
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,DigitTimeout,5
exten = 100,4,ResponseTimeout,8
exten = 100,5,BackGround(enter-conf-call-number)
exten = 100,6,Waitexten(20)
exten = 100,7,Goto(100,5)
exten = 5,1,Meetme(5)
exten = 5,2,hangup
-Original Message-
I have been beating my head against a wall trying to figure
out how I would implement a separate moderator code and
participant code for the same conference using meetme, the
deal is I dont want the participants to be able to join until
the moderator is in the conference.
Is it possible to do this using the apps as they are , or is
their a way to use an Agi script, is that the only way?
Simple extension logic is enough to do this:
From a certain extension or with a special pincode or whatever, have
moderator access. Be sure to set a database entry (/MMModerator/Roomnr/ = 1)
before accessing the MeetMe.
For all others, first check this database entry. Only access MeetMe if the
flag is set.
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RE: [Asterisk-Users] Asterisk SER (www.IPTel.org)
What would be the main benefit of this combination ? Do you expect SER to handle more registration traffic etc ? Thanks Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jiri Kuthan Sent: 03 June 2004 12:32 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk SER (www.IPTel.org) Many people use SER as proxy/registrar along with * as PSTN gateway and report it works fine. -jiri At 12:36 PM 6/3/2004, Miroslav Nachev wrote: Hi, Is it possible to use SER (www.iptel.org) toghether with Asterisk? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ 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 -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ 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] Help in direction
What does your dialplan look like. ? For sip phones, say you have ext 2001 and 2004 you would need something like ext =2001,1,Dial(SIP/2001) ext =2002,1,Dial(SIP/2002) For the phones to be able to call each other. Sorry if I am stating the obvious. Umar. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of it.albertchong.p8.hq.usSent: 02 June 2004 01:16To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Help in direction Dearall, I am new to Asterisk server. And just finish installationin Asterisk server.Server looksrunning ok.I am using X-Lite now. But I failed to connect two software phone together. Can any one give me any hints how I should troubleshooting? Can anyone give me someideas what kind of client software I should use during the test? Best regards IT Department Director of Information Technology Albert Chong 562-695-8823Ext.2201
RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
Hi Brian,
Your proposal sounds good. I think you have covered most things.
Configuration using a db would be recomended though.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian D'Arcy
Sent: 01 June 2004 21:15
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!
Spec for app_findme
Have a .conf file (findme.conf?) which contains multiple contexts, each
context's name should match the naming convention used with sip, or
iax.conf. For example, if I have [bdarcy] as one of my sip peer
entries, in findme.conf I would have, [bdarcy] also listed as an entry.
Values within each entry would be labeled something like,
[bdarcy]
ExternalNum1: 91235551212
ExternalNum2: 91235551213
etc...
app_findme would be used as the unavailable behaviour within the
dialplan (or could be used in both unavailable and busy), for example
[macro-stdexten]
exten = s,1,Wait(1)
exten = s,2,Dial(${ARG2},20,tTr)
exten = s,3,FindMe(${ARG2})
exten = s,4,Voicemail(u${ARG1})
exten = s,5,Wait(4)
exten = s,6,Hangup
exten = s,104,Voicemail(b${ARG1})
exten = s,105,Wait(2)
exten = s,106,Hangup
As the default unavailable behaviour, it always tries the findme
application, if no entries for this person exist in findme.conf, it
continues on in the dialplan, and hits the unavailable voicemail. If
entries are found:
Call gets answered, caller hears:
Hello, please wait while I try and find the person you are calling.
(MOH)
Every 10 seconds play to the caller: Still trying to find this person,
please wait..
Callee answers, app_findme says: There is a call for you from (CIDNum),
to accept this call, press *, otherwise press #, or hangup.
If I press *, the caller hears, I have found this person, connecting you
now..
Caller hears: I have found this person for you, connecting you now..
If # is pressed, the callee hangs up, or it never receives the *
confirmation tone, the caller hears: Sorry, I was unable to find this
person for you. and +101's the priority sending them into the busy
voicemail.
I look forward to hearing back from everyone on this. I'm really
excited to start learning, and feedback from the community will help
motivate me, while also ensuring I don't shelve this project just to
play some XBOX and drink some beer during my free time!
Brian D'Arcy
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RE: [Asterisk-Users] * on Opteron
I have used with Athlon 64, but noth opteron. Can imagine it being much different though. Umar -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of shabanipSent: 31 May 2004 14:55To: [EMAIL PROTECTED]Subject: [Asterisk-Users] * on Opteron anybody has success stories about running * on AMD Opteron?
RE: [Asterisk-Users] extracting country code from a number
Hi I have searched google extensively. I try not to post stuff unless I have exhausted efforts through the wiki and google. Obviously if there is something there I am not entering the right search criteria. Further help will be appreciated. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of brian k. west Sent: 30 May 2004 04:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] extracting country code from a number search google... rgagon posted something to -dev that does just this a few months back. bkw - Original Message - From: usedcanon [EMAIL PROTECTED] To: Asterisk users [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 5:01 PM Subject: [Asterisk-Users] extracting country code from a number Hi Does anyone know of an algorythm to extract the country code from a number. I understand that the country codes are of different length and there is no fixed length of local area code or phone numbers. I am sure there is a way, if not how to telephone switches handle them Umar. ___ 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] extracting country code from a number
Thank you very much indeed. Help s appreciated. (I found what I was looking for) Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Fran Boon Sent: 30 May 2004 10:40 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] extracting country code from a number On Sun, 2004-05-30 at 10:00, usedcanon wrote: Obviously if there is something there I am not entering the right search criteria. Further help will be appreciated. May 10th asterisk-dev archives. Post by Rob Gagnon: Algorithm to parse country code F ___ 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] extracting country code from a number
Hi Does anyone know of an algorythm to extract the country code from a number. I understand that the country codes are of different length and there is no fixed length of local area code or phone numbers. I am sure there is a way, if not how to telephone switches handle them Umar. ___ 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 to Asterisk - 2 question
Hi TH, Asterisk works fine as a Voicemail only server. I have it setup like that in a production setup. Configuration is simple, I will try and post something here soon. What will you integrate it with ? another asterisk system ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 28 May 2004 04:28 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to Asterisk - 2 question Hi All, I'm new to asterisk, and so far have yet to get past running the server up on a test PC. I have 2 Cisco 7960 phones to play with, both upgraded to the latest SIP image (7.1) I'd like to do 2 things, and hope that someone can point me to some simple documentation, example configs or other resources to get started: 1) A simple 1x1 setup, using the handsets described above, just to let me tinker and get an understanding of how Asterisk works. 2) A standalone voicemail server setup - Is it possible to use Asterisk just as a voicemail server ? If so, once again, any pointers to config examples etc would be appreciated. Thanks, TH ___ 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] AGI Pascal
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 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andy Powell Sent: 28 May 2004 19:32 To: [EMAIL PROTECTED] Subject: 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 ___ 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 Receptionist manager program.
Hi Kyle, I would be interested in having a look. What protocol is it using ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kyle Hagan Sent: 28 May 2004 17:33 To: Asterisk Subject: [Asterisk-Users] Asterisk Receptionist manager program. We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. We are considering writing a SIP client build into the program at a later time. Kyle ___ 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] New to Asterisk - 2 question
Hi Thomas, When I had the * server configured and running, that convinced my Boss !. How many users are you going to support. I think the main difference will be that I am using as a SIP device, you probably need to use H323 or Skinny. If you have any specific questions I would be happy to help. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 28 May 2004 09:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New to Asterisk - 2 question Umar, The plan is to integrate with a Cisco Callmanager. We currently have a very old VM system, based on a Netscape product that was installed before my time. The current project is to upgrade CM and replace the voicemail. I think Asterisk will do the job for us, now I just need to convince the boss. Any hints you can provide would be great. Thanks, Thomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of usedcanon Sent: Friday, 28 May 2004 5:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New to Asterisk - 2 question Hi TH, Asterisk works fine as a Voicemail only server. I have it setup like that in a production setup. Configuration is simple, I will try and post something here soon. What will you integrate it with ? another asterisk system ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 28 May 2004 04:28 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to Asterisk - 2 question Hi All, I'm new to asterisk, and so far have yet to get past running the server up on a test PC. I have 2 Cisco 7960 phones to play with, both upgraded to the latest SIP image (7.1) I'd like to do 2 things, and hope that someone can point me to some simple documentation, example configs or other resources to get started: 1) A simple 1x1 setup, using the handsets described above, just to let me tinker and get an understanding of how Asterisk works. 2) A standalone voicemail server setup - Is it possible to use Asterisk just as a voicemail server ? If so, once again, any pointers to config examples etc would be appreciated. Thanks, TH ___ 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] Re: FireFly doesn't work with 3rd party anymore
Hi Adam, Whats the ETA on the hardware phones. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: 28 May 2004 08:47 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore I'm going to have to go against this statement, there's one bug that I need to fix so unfortunately it will have to be Monday now. For those after the IAX/SIP firefly (albeit an old version) get http://www.virbiage.com/firefly/download/firefly-dev.exe apologies, Adam Adam Hart wrote: They'll be a new version at the end of the day (it's 9:25am now) - The reason it was like that was to cope with overlap for the firefly network going to Freshtel. Freshtel will have the Firefly Network and special version of Firefly (no IAX and SIP) while Virbiage will have a standard IAX and SIP client. Freshtel has taken our Firefly Network to allow us to concentrate on Hardware (Insert vaporware joke here) If anyone's after Australian IAX termination (or Australians wishing to call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net sorry for the dodgy version, Adam usedcanon wrote: Quite interesting, since there version history say 1.4 is the latest. The one you download is 1.7 and only works with Firefly. I have V1.5 which has the option to connect to other services. I am interested to know whats the highest version anyone has that has the other services options. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: 27 May 2004 19:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore In article [EMAIL PROTECTED], I wrote: In article [EMAIL PROTECTED], brian [EMAIL PROTECTED] wrote: Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. Are you sure? http://www.virbiage.com/firefly/download/ still says the following: Standalone SIP / IAX mode: If you want to use Firefly on our Firefly phone network (with your own voicemail etc.) then you will need to register a phone number. However, you can also use Firefly as a SIP or IAX client on your own network. Well, I just downloaded the new 1.7 build from their website (from the same page that states the above), and I see what you mean. When I first ran the new version, it still used my old settings, and successfully connected to my Asterisk server. I looked in the Options dialog, and as you say, there is no third party option at all, only the option to connect to the Firefly network. Moreover, when I changed an unrelated option (sound output device), it then overwrote my settings in the registry with new settings for the Firefly network, Freshtel. Not impressed. Especially since in their FAQ they still explicitly say it can be used with Asterisk systems. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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 ___ 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] Asterisk Receptionist manager program.
Unfortunately I am not as perfect as you and sometimes overlook the obvious. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: 28 May 2004 21:00 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk Receptionist manager program. On Fri, 2004-05-28 at 14:18, usedcanon wrote: Hi Kyle, I would be interested in having a look. What protocol is it using ? From: Kyle Hagan [EMAIL PROTECTED] We are writing a program using the manager for * Sometimes I wonder if people read at all. -- Steven Critchfield [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] Re: FireFly doesn't work with 3rd party anymore
Quite interesting, since there version history say 1.4 is the latest. The one you download is 1.7 and only works with Firefly. I have V1.5 which has the option to connect to other services. I am interested to know whats the highest version anyone has that has the other services options. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: 27 May 2004 19:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore In article [EMAIL PROTECTED], I wrote: In article [EMAIL PROTECTED], brian [EMAIL PROTECTED] wrote: Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. Are you sure? http://www.virbiage.com/firefly/download/ still says the following: Standalone SIP / IAX mode: If you want to use Firefly on our Firefly phone network (with your own voicemail etc.) then you will need to register a phone number. However, you can also use Firefly as a SIP or IAX client on your own network. Well, I just downloaded the new 1.7 build from their website (from the same page that states the above), and I see what you mean. When I first ran the new version, it still used my old settings, and successfully connected to my Asterisk server. I looked in the Options dialog, and as you say, there is no third party option at all, only the option to connect to the Firefly network. Moreover, when I changed an unrelated option (sound output device), it then overwrote my settings in the registry with new settings for the Firefly network, Freshtel. Not impressed. Especially since in their FAQ they still explicitly say it can be used with Asterisk systems. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
[Asterisk-Users] AGI Pascal
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 ___ 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
Hi Matteo, Thanks, suddenly makes sense now. I guessed that is the case however was not sure. Any opinion on what is more/most efficient, using a scripting language like perl or a compile app in C/pascal. What I am looking to do is some database access with in the script to rate a call and set an absolute time out (sort of a prepaid application, but basic with no prompts) Thanks Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brancaleoni Matteo Sent: 27 May 2004 23:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AGI Pascal Hi Il gio, 2004-05-27 alle 23:32, usedcanon ha scritto: 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. wrong wrong wrong AGI is just an external application, written in whatever language you want. if the language needs to be compiled like C, pascal, whatever... yes, must be compiled to make it work. but you are free to use scripting languages that doesn't need to be compiled... like php,perl,bash scripting,ruby, whatever AGI speaks with the app with stdout/stdin/stderr ... so anything that supports this IO can be used :) Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] Asterisk / SER or both
Hi, I am looking to implement a system to use as a prepaid service. I am aware that asterisk could do this with app_prepaid. However I am not sure if this is the best solution. Does anyone know if SER has a simmillar solution. Would I be right in assuming that SER as a SIP server is more scalable and can handle with more registerations/users ? Alternatively would it be a good idea to use SER as the SIP server and use asterisks as the PSTN gateway amd to manage the prepaid billing (given that only off net calls will be considered chargeable) Your commments and feedback will be greatly appreciated. Umar Sear ___ 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 Prepaid
Hi Steve, Sounds like more or less what I want. I would be greatful if you could send me your patch. Just wondering if you play any prompts to the user at all ? like when the credit is running out etc. Thanks Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Davies Sent: 24 May 2004 07:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Prepaid On Mon, 24 May 2004, usedcanon wrote: I have a requirement for a setup with prepaid call credits. I am aware of the two applications available (been researching for the past week), app_prepaid and app_rateengine. However neither of the two sound like exactly what I want. However I was wondering that someone who has used it might be able to say if they could be used in my scenario. Basically my scenario is pretty straight forward. Credit will be allocated to the ddi, I dont need any announcements etc (maybe low credit warning during call could be useful thoug). From the users prespective everything will be transparent. However the call should disconnect when the credit runs out. The CDR and the account DB need to be adjusted according to the call made. My guess is that app_prepaid could used with modification, I am assuming here that this is not possible as-is with configuration. Basically in case of the prepaid app, the card number can be replace transparently with the callerID. Hi, I did this to app_prepaid - you can pass a parameter into Prepaid() - its looked up in a table to find an associated card number - if that is found then the card number prompt is skipped and the associated card is used automatically. I can send a patch if you like (will also include a minor change or two to have app_prepaid work against CVS. Steve ___ 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] Asterisk Prepaid
I have a requirement for a setup with prepaid call credits. I am aware of the two applications available (been researching for the past week), app_prepaid and app_rateengine. However neither of the two sound like exactly what I want. However I was wondering that someone who has used it might be able to say if they could be used in my scenario. Basically my scenario is pretty straight forward. Credit will be allocated to the ddi, I dont need any announcements etc (maybe low credit warning during call could be useful thoug). From the users prespective everything will be transparent. However the call should disconnect when the credit runs out. The CDR and the account DB need to be adjusted according to the call made. My guess is that app_prepaid could used with modification, I am assuming here that this is not possible as-is with configuration. Basically in case of the prepaid app, the card number can be replace transparently with the callerID. All help, guidence and comments will be extremelly appreciated. Umar. ___ 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
