[asterisk-users] Queuemetrics and Asterisknow
Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- Erick Perez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenWengo + Asterisk?
2007/5/21, 0xception [EMAIL PROTECTED]: the software was unstable and crashed a lot (at least the Linux version was) ... Which version did you then use ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL/IVR Integration
Hello, I'm looking to do the following, and I wonder if Asterisk can be used for it, and if yes, if anyone can point me to the relevant information (commands, sample config...): 1. Caller dials 111, 222 or 333. 2. Based on the dialed number, Asterisk queries an external MySQL table and retrieves alphanumeric data, plays/announces it to the user and deletes the row from the database: The SQL queries would look something like: SELECT user, pwd FROM codes WHERE dialed = '111'; DELETE FROM codes WHERE user=$user AND pwd=$pwd; Thanks, David Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL/IVR Integration
David, Have a look at: http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL/IVR Integration
in 1.4, func_odbc is your friend. Julian. David wrote: Hello, I'm looking to do the following, and I wonder if Asterisk can be used for it, and if yes, if anyone can point me to the relevant information (commands, sample config...): 1. Caller dials 111, 222 or 333. 2. Based on the dialed number, Asterisk queries an external MySQL table and retrieves alphanumeric data, plays/announces it to the user and deletes the row from the database: The SQL queries would look something like: SELECT user, pwd FROM codes WHERE dialed = '111'; DELETE FROM codes WHERE user=$user AND pwd=$pwd; Thanks, David Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gustavo Souza Queiroz está ausente do escritório.
Estarei ausente do escritório a partir de 21/05/2007 e não retornarei até 11/06/2007. Responderei à sua mensagem quando retornar. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] compile asterisk in arm-linux!
hello, asteriskers: i compile asterisk 1.2.18 in arm-linux. i got this error :dlfcn.c:40: mach-o/dyld.h: No such file or directory. i check the /usr/include dir, there is no mach-o dir and dyld.h file in /usr/include. i think i am missing somethings in the cross-compile tools. Does nayone know that problem？ please give me a hint! thanks! zhulizhong - 抢注雅虎免费邮箱-3.5G容量，20M附件！ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording filename
I have figured out a way to include dialed number in recorded
voicefile in freepbx . You have to edit
/var/lib/asterisk/agi-bin/recordingcheck
add this lines after $agi=new AGI()
$temp= $agi-get_variable(DIAL_NUMBER);
$agi-verbose(Number to be dialled is -{$temp[data]});
After this you can use variable {$temp[data]} in outfile names ( set
few line below in same file ) . This is only required for freepbx .
On 30/11/06, Vicky [EMAIL PROTECTED] wrote:
No response at all :( . I did a temporary solution . I made cdr mysql to
store unique id into database from this wiki . So i now atleast have
uniquefield common in callfilename and sql records to tally .
Storing the Unique ID
Q: It would appear that the uniqueid field is not being populated in the
MySQL CDR DB. Is this an obsolete field or is a bug?
A: You need to define MYSQL_LOGUNIQUEID at compile time for it to use that
field.
You have two options in /usr/src/asterisk-addons:
1. Add CFLAGS+=-DMYSQL_LOGUNIQUEID to the Makefile.
2. Add a #define MYSQL_LOGUNIQUEID to the top of cdr_addon_mysql.c.
Finally perform the usual make clean, make, make install. Be sure to check
the Makefile for the presence of this flag after having done a CVS update!
You will most probably also want to index the uniqueid field in your cdr
table to improve performance.
On 30/11/06, Nick Hoffman [EMAIL PROTECTED] wrote:
On Wed November 29 2006 05:17, Vicky [EMAIL PROTECTED] wrote:
I am using asterisk along with freepbx . When recording is enabled for a
extension the call record file made in /var/spool/asterisk/monitor
contains information like OUT(extension
number)-(timestamp)-(uniqueid).wav . This can be a big
mess if there are
more than 1000-2000 files in that folder and very hard to locate a call
recording based on call time and extension number who dialled. I need to
put something like outgoing number dialled within call file name instead
of uniqueid .. After watching in console i opened up
/var/lib/asterisk/agi-bin/recordingcheck and saw that
it is setting
callfilename variable with extension number,time,unique id , etc. so i
edited and instead of $uniqueid i put $DIALEDPEERNUMBER ( saw in
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
) but
its just not giving dialed number and hence callfilename doesnt contain
outgoing number . Any suggestions how can i get outgoing call number in
recording file ?
Hi Vicky. Did you receive any responses to your email? I'd be interested
in
anything people suggested.
Cheers,
-- Nick
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[asterisk-users] Re: Queuemetrics and Asterisknow
I realized that queuemetrics uses Java. Is java available as an rpath package or do I need to get it from sun? Also, will it break asterisknow? Thanks. On 5/21/07, Erick Perez [EMAIL PROTECTED] wrote: Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- Erick Perez -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MySQL/IVR Integration
Func_odbc is actually also backported to 1.2, so its your friend there too. Regards Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 21. maj 2007 08:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MySQL/IVR Integration in 1.4, func_odbc is your friend. Julian. David wrote: Hello, I'm looking to do the following, and I wonder if Asterisk can be used for it, and if yes, if anyone can point me to the relevant information (commands, sample config...): 1. Caller dials 111, 222 or 333. 2. Based on the dialed number, Asterisk queries an external MySQL table and retrieves alphanumeric data, plays/announces it to the user and deletes the row from the database: The SQL queries would look something like: SELECT user, pwd FROM codes WHERE dialed = '111'; DELETE FROM codes WHERE user=$user AND pwd=$pwd; Thanks, David Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/813 - Release Date: 20-05-2007 07:54 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/813 - Release Date: 20-05-2007 07:54 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vicidial
Hi I'm looking for some help with Vicidial, If you have experience with it and could help with some consulting please contact me off list. Cheers, Joel Hill Asterisk IT [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: OK to have Asterisk and clients behind firewalls?
... but when I pick up the handset, I get no voice either way, even when I set the Linksys gateway to use a static external IP address (STUN doesn't seem to work). asterisk doesn't do STUN AFAIK, but I've never needed it and I use double NAT and have since 1.0.?. What happens when you do the echo test, call it from each phone? You don't actually need ports 1-2, a few ports for each expected channel will be enough. I have maybe 1-10020. There are a zillion settings on the phones, (which are what by the way)? Look for RTP related ones. X-Lite has a setting for xmit silence that if wrong, will not pass audio or will give one-way audio. I think the wording is do not transmit silence which should remain UNchecked. It sees do not transmit silence makes it transmit non-silence as silence :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf transcoding with asterisk
Hi, I am trying to configure asterisk to translate between rfc2833 and inband DTMF. I have a cisco gateway which is configured as a trunk, and a cisco IP phone which is registered to asterisk. The gateway does not support rfc2833 and the IP phone does. I tried changing directrtpsetup to no, and that didn't help. I tried changing canreinvite to no, but that didn't help either. I tried adding some device-specific configuaration to sip.conf, and now my calls are rejected with a status code of 404 not found. This is what I added in sip.conf: [6102] type=friend canreinvite=no host=dynamic dtmfmode=rfc2833 [trunk_1] type=peer host=192.168.20.58 canreinvite=no dtmfmode=inband What am I doing wrong? Hagai. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenWengo + Asterisk?
I believe it was a version or two ago... I just downloaded the openWango software again (current build) and it hasn't crashed on me... again I have not done any sort of extensive testing. On 5/20/07, Olivier [EMAIL PROTECTED] wrote: 2007/5/21, 0xception [EMAIL PROTECTED]: the software was unstable and crashed a lot (at least the Linux version was) ... Which version did you then use ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Users Conference this Friday: Kerry from Trixbox
Friday May 25th 2007 12:30 PM EDT Asterisk Users Live Conference/Podcast Here's a chance to ask Kerry questions about trixbox. See http://x2z.eu for access information. Listen: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 Friday June 8th 2007 Stefan Wintermeyer, author of a soon to be released asterisk book will be with us along with the English translator of his book, Stephen Bosch. The original German version has become a big seller on Amazon.de and covers 1.4. Let's put them to the torture test on the conference ;) Friday, June 1st 2007 I've asked Mark to lock some developers in the conference room until they tell us what happened at their conference. Not sure if they'll have escaped by then. What about it, Kevin? We'd love to hear from you. randulo asterisk user and enthusiast ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and fax machine
Hi, I need to share my PSTN line with my Digium card together with my FAX machine. If fax coming in, will asterisk pick up the call or my fax machine pick up the call. How do I make asterisk not to answer the incoming fax and let my fax machine receive the fax. Similarly, how do I make my fax machine not to answer any voice call and let my asterisk answer.. Regards ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Hi there, Just to announce that I've improved upon a greasemonkey script which allows users to dial any number (in the given regex format) by turning it into a clickable hyperlink. The script uses greasemonkey's ajax callback to a simple php controller script, so that the click does not navigate away from the current page. It requires an Asterisk Manager connection. See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for more details. Kind Regards, Richard Hamnett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MusicOnHold() stops after exactly 60 seconds
Hi, folks: Is there any reason why MusicOnHold() would die after 60 seconds? That looks suspiciously like a default timeout. How can I make it indefinite? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MusicOnHold() stops after exactly 60 seconds
Stephen Bosch wrote: Hi, folks: Is there any reason why MusicOnHold() would die after 60 seconds? That looks suspiciously like a default timeout. How can I make it indefinite? Moral of the story -- don't work at 4 am. The call terminates after 60 seconds because I never answered it. Changing: exten = 1234,1,MusicOnHold() to exten = 1234,1,Answer() exten = 1234,2,MusicOnHold() fixed the problem. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fax machine
Hi! Either the fax machine or the asterisk box has to pick up the call to know whether it is a fax or not. My solution is that I let asterisk pick up every call, and if it is a fax, then the call is forwarded to a fax-machine. If its a voice call, the call is forwarded to the phones. [incoming] exten = s,1,Answer() ;automatic answer for fax recognition exten = s,2,Wait(3);prevents ringing when it is a fax exten = s,3,Dial(Sip/21Sip/22Sip/25Sip/26,45,t) ;ring phones exten = s,4,Hangup ;hangup after 45 secondes ;is it a fax? then take it here! exten = fax,1,Dial(Zap/1) But this solution implies that asterisk picks up every call immediately. So the caller has to pay for the call before he can talk to you. tom aslay-pinwee wrote: Hi, I need to share my PSTN line with my Digium card together with my FAX machine. If fax coming in, will asterisk pick up the call or my fax machine pick up the call. How do I make asterisk not to answer the incoming fax and let my fax machine receive the fax. Similarly, how do I make my fax machine not to answer any voice call and let my asterisk answer.. Regards ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vicidial
Hello, Please post to the jobs/consulting forum on the VICIDIAL forums site: http://www.eflo.net/VICIDIALforum/viewforum.php?f=6 Thanks, MATT--- On 5/21/07, Joel Hill [EMAIL PROTECTED] wrote: Hi I'm looking for some help with Vicidial, If you have experience with it and could help with some consulting please contact me off list. Cheers, Joel Hill Asterisk IT [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fax machine
Hi, Thank you very much. I will test your method ASLAY - Original Message - From: Thomas Artner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 21, 2007 6:49 PM Subject: Re: [asterisk-users] asterisk and fax machine Hi! Either the fax machine or the asterisk box has to pick up the call to know whether it is a fax or not. My solution is that I let asterisk pick up every call, and if it is a fax, then the call is forwarded to a fax-machine. If its a voice call, the call is forwarded to the phones. [incoming] exten = s,1,Answer() ;automatic answer for fax recognition exten = s,2,Wait(3);prevents ringing when it is a fax exten = s,3,Dial(Sip/21Sip/22Sip/25Sip/26,45,t) ;ring phones exten = s,4,Hangup ;hangup after 45 secondes ;is it a fax? then take it here! exten = fax,1,Dial(Zap/1) But this solution implies that asterisk picks up every call immediately. So the caller has to pay for the call before he can talk to you. tom aslay-pinwee wrote: Hi, I need to share my PSTN line with my Digium card together with my FAX machine. If fax coming in, will asterisk pick up the call or my fax machine pick up the call. How do I make asterisk not to answer the incoming fax and let my fax machine receive the fax. Similarly, how do I make my fax machine not to answer any voice call and let my asterisk answer.. Regards ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMFToText Installation process
Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c , And I already tried with 'spandsp' application for this. But I am getting errors.[I followed the instructions at http://www.soft-switch.org/installing-spandsp.html] specifically by running this command:patch apps_makefile.patch I need clarification on 'ld.so.conf' file.[It has to be in the /etc/ directory. If you do not have such file - make one. In the file you need to add the path to the spandsp library.] Please give me the steps for this step. I installed asterisk 1.2.17 only, i not installed any libpri or zaptel sources. Can anybody be of help Me on this getting DTMFToText() application on asterisk with the help of app_dtmftotext.c and/or spandsp application is appreciated. Regards K.Rajesh. _ Spice up your IM conversations. New, colorful and animated emoticons. Get chatting! http://server1.msn.co.in/SP05/emoticons/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail Access
I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at ATT, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when you hear your greeting when calling yourself Toll free number What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Is there any FireFox plugin that contains an entire (SIP or IAX) softphone, that can also be scripted in the page's HTML/Javascript? On Mon, 2007-05-21 at 06:20 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 21 May 2007 10:51:09 +0100 From: Richard Hamnett [EMAIL PROTECTED] Subject: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi there, Just to announce that I've improved upon a greasemonkey script which allows users to dial any number (in the given regex format) by turning it into a clickable hyperlink. The script uses greasemonkey's ajax callback to a simple php controller script, so that the click does not navigate away from the current page. It requires an Asterisk Manager connection. See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for more details. Kind Regards, Richard Hamnett -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: OK to have Asterisk and clients behind firewalls?
I really appreciate your help :-) On Mon, 21 May 2007 10:15:40 +0200, randulo [EMAIL PROTECTED] wrote: What happens when you do the echo test, call it from each phone? Cool, I didn't know about Echo() . I added extension 111 from this example: http://www.asteriskguru.com/tutorials/echo.html Calling 111 from the remote IP phone works fine. I can hear myself. You don't actually need ports 1-2, a few ports for each expected channel will be enough. I have maybe 1-10020. Yup, I reduced them to 1-10010 on the NAT router facing Asterisk. There are a zillion settings on the phones, (which are what by the way)? I uploaded the 3102's web page here: http://codecomplete.free.fr/3102_nat/ Look for RTP related ones. X-Lite has a setting for xmit silence that if wrong, will not pass audio or will give one-way audio. I think the wording is do not transmit silence which should remain UNchecked. It sees do not transmit silence makes it transmit non-silence as silence :) From home, I tried both X-Lite and a GrandStream IP phone, both with STUN, and without opening any port on my NAT router, and they both ran the Echo() test OK. So... I guess it's something in the 3102 that must be changed so that it will finally TX/RX voice packets to remote phones (works fine when picking up an IP phone in the same LAN as the 3102 and Asterisk). Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delete voicemails after X days
Hello, I want to delete the voicemail messages that are in the Old voicemail directory, 7 days after the listening of the message by the user. Is someone as an idea how to do that??? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Really Great!!! Works for me in France I have just change the pattern and that's ok reallygood job! Cheers, Alex Richard Hamnett a écrit : Hi there, Just to announce that I've improved upon a greasemonkey script which allows users to dial any number (in the given regex format) by turning it into a clickable hyperlink. The script uses greasemonkey's ajax callback to a simple php controller script, so that the click does not navigate away from the current page. It requires an Asterisk Manager connection. See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for more details. Kind Regards, Richard Hamnett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help installing on OpenSuSE 10.2
Thanks to all that have helped me so far. I have made a lot of progress. I am able to make prilib and zaptel. Now to Asterisk... After installing the kernel source, I have: # cd /usr/src/linux # make cloneconfig # make prepare-all Then I have run ./configure in the asterisk-1.4.4 directory. I have: # make clean # make Which goes through a number of compiles and then ends up with this: asterisk2:/usr/src/asterisk-1.4.4 # make menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [LD] stereorize.o frame.o - stereorize make[1]: g++: Command not found make[1]: *** [stereorize] Error 127 make: *** [utils] Error 2 Any suggestions would be appreciated. + This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit www.tecinfo.net +___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE212P octastic initialization failure
On May 19, 2007, at 5:17 PM, Deepak Naidu wrote: I think the best way is to conact Digium Hardware support. it seems there may be an IRQ problem. No, that doesn't have anything to do with IRQ problems. It looks like it's another problem. Matthew Fredrickson -- Deepak Francois Deppierraz [EMAIL PROTECTED] wrote: Hi, I'm trying to get a TE212 working on a Dell PowerEdge 1850 running Debian etch using the latest release of libpri (1.4.0), zaptel (1.4.2.1) and asterisk (1.4.4). The initilization of the Octasic echo canceller seems to fail when the wct4xxp module is loaded. [...] VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize [...] By looking in the zaptel code, this error value (0x00103017) means cOCT6100_ERR_OPEN_EXTERNAL_MEM_BIST_FAILED. Is anyone familiar with that problem ? Thanks for your help. --- TE212P card: jumpers are set to E1 mode and nothing is connected to that card at the moment. # uname -a Linux ditti-voipa-serv-1 2.6.18-4-amd64 #1 SMP Fri May 4 00:37:33 UTC 2007 x86_64 GNU/Linux # cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 # cat /proc/interrupts CPU0 CPU1 0: 42385 0 IO-APIC-edge timer 6: 3 0 IO-APIC-edge floppy 8: 1 0 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 64 0 IO-APIC-edge ide0 169: 0 0 IO-APIC-level uhci_hcd:usb1 177: 0 0 IO-APIC-level uhci_hcd:usb2 185: 0 0 IO-APIC-level uhci_hcd:usb3 193: 19 0 IO-APIC-level ehci_hcd:usb4 201: 2148 0 IO-APIC-level ioc0 217: 1153 0 IO-APIC-level eth1 225: 160247 0 IO-APIC-level wct2xxp NMI: 64 42 LOC: 42340 42317 ERR: 0 MIS: 0 # dmesg [...] Found TE2XXP at base address fe7ffc00, remapped to c2004c00 TE2XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x7daa5400 Reg 1: 0x7daa5000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE210P (3rd Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present Failed to get chip capacity, code 0010305e! Unsupported channel capacity found on VPM module (0). Completed startup! [...] # ztcfg -v Zaptel Version: 1.4.2.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 62 channels configured. # ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who picked up with *8?
Carlos Chavez wrote: On Fri, 2007-05-18 at 16:42 -0600, Anthony Francis wrote: Use the cdr's, who wont know who but at least which phone did it. I tried following the CDR but if I dial extension 4000 and extension 4002 picks up the call using *8 the CDR says that extension 4000 ANSWERED the call. It does not say that 4002 did anything. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You will see it in the destination device field. Make sure that every sip device or line appearance has a completely unique name and parse on that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream FXS Gateway star codes
I purchased a Grandstream 4 FXS Gateway and my * extensions are
not working. I disable the special features and changed the DIAL to
{X*#+} but not luck.
I can dial any other number, receive calls and so on. This is the
only thing that seems to be an issue.
Has anybody found a way around this problem.
Reference: GXW-400x IP Analog Gateway Series
Documentation: http://www.grandstream.com/gxw400x.html
--
Yours truly,
Yu
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Re: [asterisk-users] (OT) Anyone Ever Use http://shopfort1.com as a Broker
On Sat, 2007-05-19 at 12:12 -0400, Steve Totaro wrote: I have no affiliation with them but if their quotes are accurate then they provide quite a few options as far as TDM connectivity and realtime pricing. If you do not want a phone call from a sales person, give them a BTN that goes to an IVR or something. They call no matter which box you click as far as contact me now contact me later just window shopping. I selected window shopping but they called immediately. Anyone have anything good or bad to say about this outfit. Quoted prices are really good for my needs. Much cheaper than the broker I have always used in the past. They will not tell you the carrier until you speak with a sales person but It's all Verizon (tm). Thanks, Steve Totaro http://www.asteriskhelpdesk.com Steve, We have been a ShopforT1 dealer for over 5 years. The pricing is solid, service is quick, as for knowing the carrier before you buy, that is easy, we can and will produce quotes detailing the carriers for you. As for the remainder, actually we handle all the paperwork filing for the carrier, so when it comes time to sign, you deal directly with us as the carrier. Support and installation is provided directly by the carrier, so there is no possible downfall there, we act solely as the carriers direct agent to you. Give me a shout off list if you would like to discuss it further. James- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Re install
I had to re install the my Asterisk BE with the latest version, and when I try to load my g.729 codec license I do not see the folders in the path that they are described in the instructions given to us with the license or in your online documentation. I installed the disk 1 immage (rPath), and I am not able to perform the g.729 installation or registration. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fax machine
Another solution: http://www.vikingtelecomsolutions.com/catalog/model_FAXJ-300.htm?sid=046EBF6027C7A0D38E77EAF75B184540pid=1209 Jorge aslay-pinwee wrote: Hi, Thank you very much. I will test your method ASLAY - Original Message - From: Thomas Artner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 21, 2007 6:49 PM Subject: Re: [asterisk-users] asterisk and fax machine Hi! Either the fax machine or the asterisk box has to pick up the call to know whether it is a fax or not. My solution is that I let asterisk pick up every call, and if it is a fax, then the call is forwarded to a fax-machine. If its a voice call, the call is forwarded to the phones. [incoming] exten = s,1,Answer() ;automatic answer for fax recognition exten = s,2,Wait(3);prevents ringing when it is a fax exten = s,3,Dial(Sip/21Sip/22Sip/25Sip/26,45,t) ;ring phones exten = s,4,Hangup ;hangup after 45 secondes ;is it a fax? then take it here! exten = fax,1,Dial(Zap/1) But this solution implies that asterisk picks up every call immediately. So the caller has to pay for the call before he can talk to you. tom aslay-pinwee wrote: Hi, I need to share my PSTN line with my Digium card together with my FAX machine. If fax coming in, will asterisk pick up the call or my fax machine pick up the call. How do I make asterisk not to answer the incoming fax and let my fax machine receive the fax. Similarly, how do I make my fax machine not to answer any voice call and let my asterisk answer.. Regards ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fax machine
On Mon, 2007-05-21 at 12:49 +0200, Thomas Artner wrote: My solution is that I let asterisk pick up every call, and if it is a fax, then the call is forwarded to a fax-machine. If its a voice call, the call is forwarded to the phones. That is what I do as well. Use the fax extension to forward FAX calls. This does require that the FAX machine be connected to an FXS port on your Digium card. I do this with another computer with a FAX modem connected to an FXS port on my Digium card and it works great. Eventually, I'd like to move the FAX modem to the same machine with the Digium card so that I won't have to have another machine up to receive faxes; anybody ever tried that? It would be something I've never tried, having an RJ-11 cable between two cards in the same box. Any grounding/feedback issues with doing that? --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Matthew Rubenstein a écrit : Is there any FireFox plugin that contains an entire (SIP or IAX) softphone, that can also be scripted in the page's HTML/Javascript? Have you looked at MozPhone (http://moziax.mozdev.org/) ? It's a Firefox VoIP extension IAX softphone, and Asterisk manager interface. It does include click to dial, click to transfer, and could do more from a web page through javascript. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FW: Re install
I was able to fid the modules directoty, but when I run -r-x-- 1 root root 1288344 May 21 11:35 register /root/register I get the following error -bash: /root/register: cannot execute binary file I have changed the file attributes as you can see on the ls -l From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Monday, May 21, 2007 11:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Re install I had to re install the my Asterisk BE with the latest version, and when I try to load my g.729 codec license I do not see the folders in the path that they are described in the instructions given to us with the license or in your online documentation. I installed the disk 1 immage (rPath), and I am not able to perform the g.729 installation or registration. image001.png___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAS signalling conflicts with Clear channel
Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. As I try to initialize zap I get a CAS signalling conflict (see below). What does it mean? Since it refers to span 2 could it be that the second digital card (B410P) is interfering somehow? How can I avoid this conflict (I will of course try to remove the B410P card but right now I don't have access to the server)? Help greatly appreciated. # modprobe -a zaptel # modprobe -a wcte12xp # ztcfg -v Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 101: FXS Kewlstart (Default) (Slaves: 101) Channel 102: FXS Kewlstart (Default) (Slaves: 102) Channel 103: FXS Kewlstart (Default) (Slaves: 103) Channel 104: FXS Kewlstart (Default) (Slaves: 104) 35 channels configured. Changing signalling on channel 1 from FXS Kewlstart to Clear channel Changing signalling on channel 2 from FXS Kewlstart to Clear channel Changing signalling on channel 3 from FXS Kewlstart to Clear channel Changing signalling on channel 4 from FXS Kewlstart to Clear channel Changing signalling on channel 5 from Unused to Clear channel Changing signalling on channel 6 from Unused to Clear channel Changing signalling on channel 7 from Unused to Clear channel Changing signalling on channel 8 from Unused to Clear channel Changing signalling on channel 9 from Unused to Clear channel Changing signalling on channel 10 from Unused to Clear channel Changing signalling on channel 11 from Unused to Clear channel Changing signalling on channel 12 from Unused to Clear channel Changing signalling on channel 13 from Unused to Clear channel Changing signalling on channel 14 from Unused to Clear channel Changing signalling on channel 15 from Unused to Clear channel Changing signalling on channel 16 from Unused to HDLC with FCS check Changing signalling on channel 17 from Unused to Clear channel Changing signalling on channel 18 from Unused to Clear channel Changing signalling on channel 19 from Unused to Clear channel CAS signalling on span 2 conflicts with Clear channel on channel 20. # cat /etc/zaptel.conf # WCTDM/0 Wildcard TDM400P REV I Board 1 fxsks=101 fxsks=102 fxsks=103 fxsks=104 # TE120P (PRI): span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone= es defaultzone = es # cat /etc/asterisk/zapata.conf [channels] context=from-pstn signalling = fxs_ks cidsignalling=dtmf busydetect=yes callprogress=no group=0 faxdetect=both channel=101-104 switchtype = euroisdn signalling = pri_cpe context=from-pstn group = 1 channel = 1-15,17-31 # lspci -vb 00:00.0 Host bridge: Broadcom GCNB-LE Host Bridge (rev 01) Flags: fast devsel 00:00.1 Host bridge: Broadcom GCNB-LE Host Bridge Flags: fast devsel 00:02.0 Ethernet controller: Intel Corporation 82540EM Gigabit Ethernet Controller (rev 02) Subsystem: Dell PowerEdge 600SC Flags: bus master, 66MHz, medium devsel, latency 32, IRQ 10 Memory at fe10 (32-bit, non-prefetchable) I/O ports at ecc0 Capabilities: [dc] Power Management version 2 Capabilities:
Re: [asterisk-users] Re: Queuemetrics and Asterisknow
Hello Erick, I believe that if you go for a manual installation of non-AsteriskNOW components (like Java) they should be excluded from the components that Conary mantains. l. On Mon, 21 May 2007 09:54:52 +0200, Erick Perez [EMAIL PROTECTED] wrote: I realized that queuemetrics uses Java. Is java available as an rpath package or do I need to get it from sun? Also, will it break asterisknow? Thanks. On 5/21/07, Erick Perez [EMAIL PROTECTED] wrote: Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- Erick Perez -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAS signalling conflicts with Clear channel
On 5/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. I could be wrong here, but I don't think you get to arbitrarily make up what the channel numbers. At least I've never done that; I let the first channel be 1, second one 2, etc, through all of the cards, based on loading order of the PCI cards. And are you sure about the loading order of the cards? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Anyone Ever Use http://shopfort1.com as a Broker
On Sat, 2007-05-19 at 12:12 -0400, Steve Totaro wrote: I have no affiliation with them but if their quotes are accurate then they provide quite a few options as far as TDM connectivity and realtime pricing. If you do not want a phone call from a sales person, give them a BTN that goes to an IVR or something. They call no matter which box you click as far as contact me now contact me later just window shopping. I selected window shopping but they called immediately. Anyone have anything good or bad to say about this outfit. Quoted prices are really good for my needs. Much cheaper than the broker I have always used in the past. They will not tell you the carrier until you speak with a sales person but It's all Verizon (tm). Thanks, Steve Totaro http://www.asteriskhelpdesk.com Steve, We have been a ShopforT1 dealer for over 5 years. The pricing is solid, service is quick, as for knowing the carrier before you buy, that is easy, we can and will produce quotes detailing the carriers for you. As for the remainder, actually we handle all the paperwork filing for the carrier, so when it comes time to sign, you deal directly with us as the carrier. Support and installation is provided directly by the carrier, so there is no possible downfall there, we act solely as the carriers direct agent to you. Give me a shout off list if you would like to discuss it further. James- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI: Not Found. Move along
Did you acess the page at port 8088 ? I.e.: http://192.168.0.1:8088 On 5/17/07, Tim Verscheure [EMAIL PROTECTED] wrote: Still nothing. I'll give my config files: manager.conf ; ; Asterisk Call Management support ; ; By default asterisk will listen on localhost only. [general] displaysystemname = yes enabled = yes webenabled = yes port = 5038 httptimeout = 60 bindaddr = 0.0.0.0 ; No access is allowed by default. ; To set a password, create a file in /etc/asterisk/manager.d ; use creative permission games to allow other serivces to create their own ; files #include manager.d/*.conf [admin] secret = javali deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 permit=192.168.1.68/255.255.255.0 read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config [panel] secret = javali deny=0.0.0.0/0.0.0.0 permit=192.168.1.68/255.255.255.0 permit=127.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config http.conf --- ; ; Asterisk Builtin mini-HTTP server ; ; [general] ; ; Whether HTTP interface is enabled or not. Default is no. ; enabled=yes ; ; Whether Asterisk should serve static content from http-static ; Default is no. ; enablestatic=yes ; ; Address to bind to. Default is 0.0.0.0 ; bindaddr=0.0.0.0 ; ; Port to bind to (default is 8088) ; bindport=8088 ; ; Prefix allows you to specify a prefix for all requests ; to the server. The default is asterisk so that all ; requests must begin with /asterisk ; ;prefix=asterisk ; The post_mappings section maps URLs to real paths on the filesystem. If a ; POST is done from within an authenticated manager session to one of the ; configured POST mappings, then any files in the POST will be placed in the ; configured directory. ; ;[post_mappings] ; ; In this example, if the prefix option is set to asterisk, then using the ; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/. ;uploads = /var/lib/asterisk/uploads/ ; thanks in advance, Tim 2007/5/17, Troy Ayers [EMAIL PROTECTED]: Tim Verscheure wrote: Hi there, I just installed the GUI for Asterisk 1.4.4 and correctly set my settings but when I use my browser to access it, it gives me an error saying Not Found. Nothing to see here, move along with asterisk in the header and footer... anyone had this problemn before? greetz Try https:// not http:// -Troy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes MSN:[EMAIL PROTECTED] (48) 99115299 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAS signalling conflicts with Clear channel
--- David Gomillion [EMAIL PROTECTED] wrote: On 5/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. I could be wrong here, but I don't think you get to arbitrarily make up what the channel numbers. At least I've never done that; I let the first channel be 1, second one 2, etc, through all of the cards, based on loading order of the PCI cards. And are you sure about the loading order of the cards? I'm sure you're right because the following yields no error: # misdn-init stop # rmmod wctdm # rmmod xpp # rmmod wcte12xp # rmmod zaptel # modprobe -a zaptel # modprobe -a wcte12xp # ztcfg -v Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to Clear channel Changing signalling on channel 4 from Unused to Clear channel Changing signalling on channel 5 from Unused to Clear channel Changing signalling on channel 6 from Unused to Clear channel Changing signalling on channel 7 from Unused to Clear channel Changing signalling on channel 8 from Unused to Clear channel Changing signalling on channel 9 from Unused to Clear channel Changing signalling on channel 10 from Unused to Clear channel Changing signalling on channel 11 from Unused to Clear channel Changing signalling on channel 12 from Unused to Clear channel Changing signalling on channel 13 from Unused to Clear channel Changing signalling on channel 14 from Unused to Clear channel Changing signalling on channel 15 from Unused to Clear channel Changing signalling on channel 16 from Unused to HDLC with FCS check Changing signalling on channel 17 from Unused to Clear channel Changing signalling on channel 18 from Unused to Clear channel Changing signalling on channel 19 from Unused to Clear channel Changing signalling on channel 20 from Unused to Clear channel Changing signalling on channel 21 from Unused to Clear channel Changing signalling on channel 22 from Unused to Clear channel Changing signalling on channel 23 from Unused to Clear channel Changing signalling on channel 24 from Unused to Clear channel Changing signalling on channel 25 from Unused to Clear channel Changing signalling on channel 26 from Unused to Clear channel Changing signalling on channel 27 from Unused to Clear channel Changing signalling on channel 28 from Unused to Clear channel Changing signalling on channel 29 from Unused to Clear channel Changing signalling on channel 30 from Unused to Clear channel Changing signalling on channel 31 from Unused to Clear channel I guess I'll have trouble getting all three cards to work together on the same box. Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/features_spam.html
Re: [asterisk-users] GUI: Not Found. Move along
yes!! 2007/5/21, Guilherme Góes [EMAIL PROTECTED]: Did you acess the page at port 8088 ? I.e.: http://192.168.0.1:8088 On 5/17/07, Tim Verscheure [EMAIL PROTECTED] wrote: Still nothing. I'll give my config files: manager.conf ; ; Asterisk Call Management support ; ; By default asterisk will listen on localhost only. [general] displaysystemname = yes enabled = yes webenabled = yes port = 5038 httptimeout = 60 bindaddr = 0.0.0.0 ; No access is allowed by default. ; To set a password, create a file in /etc/asterisk/manager.d ; use creative permission games to allow other serivces to create their own ; files #include manager.d/*.conf [admin] secret = javali deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 permit=192.168.1.68/255.255.255.0 read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config [panel] secret = javali deny=0.0.0.0/0.0.0.0 permit=192.168.1.68/255.255.255.0 permit=127.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config http.conf --- ; ; Asterisk Builtin mini-HTTP server ; ; [general] ; ; Whether HTTP interface is enabled or not. Default is no. ; enabled=yes ; ; Whether Asterisk should serve static content from http-static ; Default is no. ; enablestatic=yes ; ; Address to bind to. Default is 0.0.0.0 ; bindaddr=0.0.0.0 ; ; Port to bind to (default is 8088) ; bindport=8088 ; ; Prefix allows you to specify a prefix for all requests ; to the server. The default is asterisk so that all ; requests must begin with /asterisk ; ;prefix=asterisk ; The post_mappings section maps URLs to real paths on the filesystem. If a ; POST is done from within an authenticated manager session to one of the ; configured POST mappings, then any files in the POST will be placed in the ; configured directory. ; ;[post_mappings] ; ; In this example, if the prefix option is set to asterisk, then using the ; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/. ;uploads = /var/lib/asterisk/uploads/ ; thanks in advance, Tim 2007/5/17, Troy Ayers [EMAIL PROTECTED]: Tim Verscheure wrote: Hi there, I just installed the GUI for Asterisk 1.4.4 and correctly set my settings but when I use my browser to access it, it gives me an error saying Not Found. Nothing to see here, move along with asterisk in the header and footer... anyone had this problemn before? greetz Try https:// not http:// -Troy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes MSN:[EMAIL PROTECTED] (48) 99115299 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call someone to instantly join conference using MeetMe
Hi Ya that works good. Thanks Arpit On 5/20/07, Kapil Dhawan [EMAIL PROTECTED] wrote: Arpit Use Auto dial. http://www.voip-info.org/wiki-Asterisk+auto-dial+out Create a .call file as mentioned by Dave. Dave Miller wrote: Arpit Mehta wrote on 5/19/07 10:18 PM: I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or any other conference application? Any suggestions/hints/links are welcome. Set up an extension that dials directly into the conference in question, then use that extension via the Local channel as the source of a call to the number you want to dial, triggered via the Management API or a call file. [meetme-dialin] exten = 1234,1,Answer() exten = 1234,n,MeetMe(4321) Pipe the following into the Manager API with an extra blank line at the end: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: from-inside (or whatever context is appropriate) Exten: (the number you want to call) Priority: 1 I'm going from memory, so you may have to play with it a little bit but that's the basic idea. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAS signalling conflicts with Clear channel
Vieri, Make sure you are loading the digital card first and then analog card. I had the same problem and Digium engineers helped me out. Cheers, Nitesh Vieri wrote: --- David Gomillion [EMAIL PROTECTED] wrote: On 5/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. I could be wrong here, but I don't think you get to arbitrarily make up what the channel numbers. At least I've never done that; I let the first channel be 1, second one 2, etc, through all of the cards, based on loading order of the PCI cards. And are you sure about the loading order of the cards? I'm sure you're right because the following yields no error: # misdn-init stop # rmmod wctdm # rmmod xpp # rmmod wcte12xp # rmmod zaptel # modprobe -a zaptel # modprobe -a wcte12xp # ztcfg -v Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to Clear channel Changing signalling on channel 4 from Unused to Clear channel Changing signalling on channel 5 from Unused to Clear channel Changing signalling on channel 6 from Unused to Clear channel Changing signalling on channel 7 from Unused to Clear channel Changing signalling on channel 8 from Unused to Clear channel Changing signalling on channel 9 from Unused to Clear channel Changing signalling on channel 10 from Unused to Clear channel Changing signalling on channel 11 from Unused to Clear channel Changing signalling on channel 12 from Unused to Clear channel Changing signalling on channel 13 from Unused to Clear channel Changing signalling on channel 14 from Unused to Clear channel Changing signalling on channel 15 from Unused to Clear channel Changing signalling on channel 16 from Unused to HDLC with FCS check Changing signalling on channel 17 from Unused to Clear channel Changing signalling on channel 18 from Unused to Clear channel Changing signalling on channel 19 from Unused to Clear channel Changing signalling on channel 20 from Unused to Clear channel Changing signalling on channel 21 from Unused to Clear channel Changing signalling on channel 22 from Unused to Clear channel Changing signalling on channel 23 from Unused to Clear channel Changing signalling on channel 24 from Unused to Clear channel Changing signalling on channel 25 from Unused to Clear channel Changing signalling on channel 26 from Unused to Clear channel Changing signalling on channel 27 from Unused to Clear channel Changing signalling on channel 28 from Unused to Clear channel Changing signalling on channel 29 from Unused to Clear channel Changing signalling on channel 30 from Unused to Clear channel Changing signalling on channel 31 from Unused to Clear channel I guess I'll have trouble getting all three cards to work together on the same box.
Re: [asterisk-users] CAS signalling conflicts with Clear channel
On 5/21/07, Vieri [EMAIL PROTECTED] wrote: --- David Gomillion [EMAIL PROTECTED] wrote: On 5/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. I could be wrong here, but I don't think you get to arbitrarily make up what the channel numbers. At least I've never done that; I let the first channel be 1, second one 2, etc, through all of the cards, based on loading order of the PCI cards. And are you sure about the loading order of the cards? I'm sure you're right because the following yields no error: # misdn-init stop # rmmod wctdm # rmmod xpp # rmmod wcte12xp # rmmod zaptel # modprobe -a zaptel # modprobe -a wcte12xp # ztcfg -v snip I guess I'll have trouble getting all three cards to work together on the same box. You should still be able to get all of the cards working together. Just be sure you define your channels in the right order. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra MWI
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail Access
I created a *9 extension which executes VoiceMailMain with the callerid number as the argument. Then of course the voicemail box just has to be the same as the phone number. Then we just have another DID for outside access. * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Mike Hammett wrote: I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at ATT, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when you hear your greeting when calling yourself Toll free number What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI: Festival Ringing on Screening not working properly
I am running into two problems:
1) The ringing stops during call screening once the extension picks up (but
has not yet approved call)
When a person calls and choose an extension, the Dial link is called and the
person hears the ring -- but as soon as the receiving caller picks up (even
though they have to approved the call), the ringing stops... the person
calling hears silence.
This is obviously undesirable as it will confuse the person calling in...
2) Festival will not work during screening -- I want to use festival to
announce a message but it plays nothing and the macro-screen seems to
terminate immediately and connect the caller. I have tested that the
Festival command does work in a normal situation (I put the line earlier
before AGI is called without any problem)
Oh this is under Asterisk 1.4.4
Thanks in advance,
-John
Code:
AGI script executes
==
my $res = $AGI-exec(Dial,SIP/$call|20|rM(screen));
where $call is a sip extension..
and the following macro in extensions.conf
===
[macro-screen]
exten = s,1,Wait(1)
exten = s,n,Read(ACCEPT|initialGreeting|1)
exten = s,n,Set(MACRO_RESULT=CONTINUE)
exten = s,n,GotoIf($[${ACCEPT} = 1 ] ?60:50)
exten = s,50,NoOp(1 not pressed)
exten = s,n,Hangup()
exten = s,60,Set(MACRO_RESULT=)
exten = s,n,NoOp(Done)
Above code works fine other than ringing issue but if I add this line after
Wait then you hear nothing and it bombs out:
exten = s,n,Festival('You have received a business call')
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[asterisk-users] MoH WAY too loud
Hi folks! I'm having a problem where my music on hold is just blaring to my callers. I've tried several different formats (converting using mpg123 and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail. Every file plays way too loud. I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what parameter to pass. Any help any of you can provide would be much appreciated, thanks. Jay PS - What file type should I be using for MoH anyway? I know mp3 is out, but is wav or gsm preferred? Or is there another format I should consider? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who picked up with *8?
Anthony Francis wrote: Carlos Chavez wrote: On Fri, 2007-05-18 at 16:42 -0600, Anthony Francis wrote: Use the cdr's, who wont know who but at least which phone did it. I tried following the CDR but if I dial extension 4000 and extension 4002 picks up the call using *8 the CDR says that extension 4000 ANSWERED the call. It does not say that 4002 did anything. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You will see it in the destination device field. Make sure that every sip device or line appearance has a completely unique name and parse on that. Make that the Destination Channel, because the channel name will always have the sip name in it if the call was terminated by that asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Cool, please send me the pattern so i can add it Cheers Rick On 5/21/07, Alexandre VERNIOL [EMAIL PROTECTED] wrote: Really Great!!! Works for me in France I have just change the pattern and that's ok reallygood job! Cheers, Alex Richard Hamnett a écrit : Hi there, Just to announce that I've improved upon a greasemonkey script which allows users to dial any number (in the given regex format) by turning it into a clickable hyperlink. The script uses greasemonkey's ajax callback to a simple php controller script, so that the click does not navigate away from the current page. It requires an Asterisk Manager connection. See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for more details. Kind Regards, Richard Hamnett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail Access
Mike Hammett wrote:
I was looking at the ILECs’ web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during the greeting to
enter the voicemail system? If they call their own number, how do I get
Asterisk to recognize that and take them to the voicemail system?
Mike,
A common approach is to use the caller id in combination with some digit
sequence. For my systems, I've just used 555 as the VM extension.
exten=555,1,VoicemailMain(${CALLERID(num)})
For access to the VM from outside the system, I've used an AGI script to
query a database to validate the user.
--
Warm Regards,
Lee
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Re: [asterisk-users] CAS signalling conflicts with Clear channel
Thanks Nitesh, I did just that and got both the TE120P PRI and the analog card working together. The 4-BRI mISDN B410P is a bit tougher and I still haven't understood yet where the channels are supposed to be specified (if so) in /etc/asterisk/misdn.conf. I've used www.misdn.org , NOT the included make b410p in zaptel's package. Has anyone successfully configured a B410P with mISDN from www.misdn.org (hfcmulti)? If so, can you share your relevant config files? --- Nitesh Divecha [EMAIL PROTECTED] wrote: Vieri, Make sure you are loading the digital card first and then analog card. I had the same problem and Digium engineers helped me out. Cheers, Nitesh Vieri wrote: --- David Gomillion [EMAIL PROTECTED] wrote: On 5/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed them to 101-104. I could be wrong here, but I don't think you get to arbitrarily make up what the channel numbers. At least I've never done that; I let the first channel be 1, second one 2, etc, through all of the cards, based on loading order of the PCI cards. And are you sure about the loading order of the cards? I'm sure you're right because the following yields no error: # misdn-init stop # rmmod wctdm # rmmod xpp # rmmod wcte12xp # rmmod zaptel # modprobe -a zaptel # modprobe -a wcte12xp # ztcfg -v Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemails after X days
You could schedule a cron job to run a shell script to delete any files in the //voicemail/*/Old/ directory that are older than the amount of time specified. You could craft something up by comparing the date modification timestamp from `ls -l` or the access modification from `ls -lu`(?). I don't know of any Asterisk features to delete the older voicemail. -kn0x On 5/21/07, David Florella [EMAIL PROTECTED] wrote: Hello, I want to delete the voicemail messages that are in the *Old * voicemail directory, 7 days after the listening of the message by the user. Is someone as an idea how to do that??? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra MWI
On Mon, 2007-05-21 at 14:39 -0400, Lee Jenkins wrote: I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! In my experience it is never necessary to set the Explicit MWI, all the Aastra phones turn on the message light with the default configuration. Could you show your sip.conf definition for the phone? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: DUNDi configuration problem
Now I get this... If I call from 5011 on the 192.168.1.103 machine to 6010 on the 192.168.1.69 machine my X-lite softphone says, call declined this is the output: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508, ext-local|6010|1) in new stack -- Goto (ext-local,6010,1) [May 21 15:32:46] WARNING[8939]: pbx.c:2450 __ast_pbx_run: Channel 'SIP/5011-081da508' sent into invalid extension '6010' in context 'ext-local', but no invalid handler I'll add my extension file so you can see it. greetz 2007/5/19, Remco Post [EMAIL PROTECTED]: Tim Verscheure wrote: If I read all this is realize what a noob I am in this matter. Could I make a call by saying something like this: exten = 16000,1,Dial(SIP/[EMAIL PROTECTED]) you could, look into the DUNDILOOKUP function... Or something like that? 2007/5/19, Remco Post [EMAIL PROTECTED]: Tim Verscheure wrote: like this??? [dundi-priv-switch] ; Just a wrapper for the switch switch = DUNDi/priv yes that should do. Does your asterisk console show anything useful? And if you do wind up in the switch, what does you dundi debug show? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users extensions.conf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MoH WAY too loud
Jay Moore wrote: Hi folks! I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what parameter to pass. Doing a 'man sox' does wonders: -v volume Change amplitude (floating point); less than 1.0 decreases, greater than 1.0 increases. May use a negative number to invert the phase of the audio data. It is interesting to note that we perceive volume logarithmically but this adjusts the amplitude linearly. So, this is how I increase the volume on my paging sox paging.gsm -v 4 /var/lib/asterisk/sounds/outx2.gsm Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate and bridge Can it be done? Best Way?
Hi, Im new, but trying real hard! I just need general direction, not details yet..i'll try to figure those...just looking to avoid brick walls...bottlenecks...inefficiencies etc upfront. Hardware: motorola vt2442 - trixbox Apps: Dot Net application that operates the Manager API and the FASTAGI interfaces. I have the 2442 set as a PLAR so as soon as the ext is off-hook, it dials into the *61 dial-plan. This is what I want to happen: Ext goes off-hook vt2442 PLAR auto-dials *61 *61 dial-plan gets some info from the inside user Need help for everything below: inside user is put on hold..queued or parked or put in a conf??? Sep call is originated to an outside line If call answered some info is gathered from the outside party outside party is joined to inside party. I am willing to use any or all of a combo of the following: Manager API FASTAGI dial-plans .call files or anything else!! The Manager/FastAGI application can perform any action required. thanks in advance! -Henry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra MWI
Lee Jenkins wrote: I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Did you set the mailbox= variable in sip.conf? I made that mistake yesterday and wasted 90 minutes hunting for the problem. -s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MoH WAY too loud
Doug, Thanks for the reply. Immediately after hitting send I found exactly what I was looking for. Don't know why I didn't consider doing a 'man sox' earlier. I must be getting senile. ;) That said, I altered my initial .gsm files and made them 75% quieter (-v .25). I replaced my loud files with my newer, quieter files and reloaded res_musiconhold.so to no avail. I confirmed the new files *are* quieter, but Asterisk still plays them extremely loud. Do I need to reload a different module, or perhaps completely restart Asterisk to use these newer files? Thanks, Jay Doug Lytle wrote: Jay Moore wrote: Hi folks! I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what parameter to pass. Doing a 'man sox' does wonders: -v volume Change amplitude (floating point); less than 1.0 decreases, greater than 1.0 increases. May use a negative number to invert the phase of the audio data. It is interesting to note that we perceive volume logarithmically but this adjusts the amplitude linearly. So, this is how I increase the volume on my paging sox paging.gsm -v 4 /var/lib/asterisk/sounds/outx2.gsm Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help installing on OpenSuSE 10.2
Malcom Kemp wrote: make[1]: g++: Command not found hint :) -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: DUNDi configuration problem
Tim Verscheure wrote: Now I get this... If I call from 5011 on the 192.168.1.103 machine to 6010 on the 192.168.1.69 machine my X-lite softphone says, call declined this is the output: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508, ext-local|6010|1) in new stack -- Goto (ext-local,6010,1) [May 21 15:32:46] WARNING[8939]: pbx.c:2450 __ast_pbx_run: Channel 'SIP/5011-081da508' sent into invalid extension '6010' in context 'ext-local', but no invalid handler so, is there an extension 6010 in you context ext-local? Probably not ;-) I'll add my extension file so you can see it. greetz 2007/5/19, Remco Post [EMAIL PROTECTED]: Tim Verscheure wrote: If I read all this is realize what a noob I am in this matter. Could I make a call by saying something like this: exten = 16000,1,Dial(SIP/[EMAIL PROTECTED]) you could, look into the DUNDILOOKUP function... Or something like that? 2007/5/19, Remco Post [EMAIL PROTECTED]: Tim Verscheure wrote: like this??? [dundi-priv-switch] ; Just a wrapper for the switch switch = DUNDi/priv yes that should do. Does your asterisk console show anything useful? And if you do wind up in the switch, what does you dundi debug show? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] getting a call back from voicemail?
Hi, is there a way or feature available in Asterisk where one can 'pull' a call back from voicemail. i.e. if you don't get to the phone in time and it goes to voicemail, can you key some sequence in and pull the caller out of voicemail and speak to them? Thanks Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting a call back from voicemail?
Mike Dent wrote: Hi, is there a way or feature available in Asterisk where one can 'pull' a call back from voicemail. i.e. if you don't get to the phone in time and it goes to voicemail, can you key some sequence in and pull the caller out of voicemail and speak to them? It seems like you should be able to transfer the caller's channel to another extension.. That extension would ring, though, so it wouldn't be an immediate connect. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Windows Media streaming for MOH?
Anyone have Windows Media streaming for MOH working? I followed the various procedures on the Asterisk Wiki for using mplayer which seems to be the only Linux player capable of playing windows media streaming audio (asf, wmv etc.). Anyone get this working? I can get shoutcast streams working using mpg123 but so far no luck with windows media streaming. Is there another player out there or a trick of some sort? I've been googling but so far no luck. The problem is that many radio stations including the local ones people in my area use for MOH on their traditional PBX's use windows media streaming. As a work around I would consider streaming it from a softphone on a Windows PC to a conference room if I had to. That may be easier to do but haven't found much info on that either. Worst case I would consider a receiver on Line In on a softphone on a Windows PC or absolute worst case, from line in audio on the Asterisk server. Streaming local radio directly on the Asterisk server is the most elegant solution IMHO. Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra MWI
Stephen Bosch wrote: Lee Jenkins wrote: I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Did you set the mailbox= variable in sip.conf? I made that mistake yesterday and wasted 90 minutes hunting for the problem. -s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Make sure you set it as a literal just like you would send the call to voicemail. i.e. [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Windows Media streaming for MOH?
shadowym wrote: Anyone have Windows Media streaming for MOH working? I followed the various procedures on the Asterisk Wiki for using mplayer which seems to be the only Linux player capable of playing windows media streaming audio (asf, wmv etc.). Anyone get this working? I can get shoutcast streams working using mpg123 but so far no luck with windows media streaming. Is there another player out there or a trick of some sort? I've been googling but so far no luck. The problem is that many radio stations including the local ones people in my area use for MOH on their traditional PBX's use windows media streaming. As a work around I would consider streaming it from a softphone on a Windows PC to a conference room if I had to. That may be easier to do but haven't found much info on that either. Worst case I would consider a receiver on Line In on a softphone on a Windows PC or absolute worst case, from line in audio on the Asterisk server. Streaming local radio directly on the Asterisk server is the most elegant solution IMHO. Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Easiest way without a native decoder is to have another machine receive the wmv and transcode it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra MWI
Stephen Bosch wrote: Lee Jenkins wrote: I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Did you set the mailbox= variable in sip.conf? I made that mistake yesterday and wasted 90 minutes hunting for the problem. Hey guys, Thanks for responding. Yes, I have set that. Here is my sip.conf entry for that extension: [117] context=Management type=friend canreinvite=no dtmfmode=rfc2833 callerid=Napoleon Hill302-539- nat=no port=5060 qualify=no secrete=117 host=192.168.1.117 [EMAIL PROTECTED] disallow=all allow-g729 allow=ulaw -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra MWI
Lee Jenkins wrote: Stephen Bosch wrote: Lee Jenkins wrote: I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Did you set the mailbox= variable in sip.conf? I made that mistake yesterday and wasted 90 minutes hunting for the problem. Hey guys, Thanks for responding. Yes, I have set that. Here is my sip.conf entry for that extension: [117] context=Management type=friend canreinvite=no dtmfmode=rfc2833 callerid=Napoleon Hill302-539- nat=no port=5060 qualify=no secrete=117 host=192.168.1.117 [EMAIL PROTECTED] disallow=all allow-g729 allow=ulaw is this actually autheticating? secrete=117 should be secret=117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MoH WAY too loud
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jay Moore wrote: Hi folks! I'm having a problem where my music on hold is just blaring to my callers. I've tried several different formats (converting using mpg123 and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail. Every file plays way too loud. What are you using for incoming calls? If ZAP, you sure there's not too much gain there? Maybe the phones are quiet and the line is turned up. Do you get this when you call MOH internally? - -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGUg/ADQNt8rg0Kp4RAtUPAJ0Wuw0b6UiEJhuzY7phz0RGlIz1YwCeMiWF HMRZlqWq9w4EEcfiEgwJMoA= =Q/yM -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Aastra MWI
This is probably cold comfort but I have NEVER had any issues with MWI working on Aastra phones. It always just works by default. No extra configuration necessary on the phone for sure. Just reset it to factory defaults. Explicit MWI is NOT checked by default and I have never had to check it. No extra configuration on Freepbx/Trixbox. Not sure about a basic Asterisk install but here is my sip.conf. [general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying The number you have dialed is not in service. Please check the ; number and try again. context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf [600] type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=no [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all dial=SIP/600 context=from-internal canreinvite=no callerid=device 600 allow=ulaw -Original Message- From: Lee Jenkins [mailto:[EMAIL PROTECTED] Sent: Monday, May 21, 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Aastra MWI I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate and bridge Can it be done? Best Way?
Henry wrote: Hi, Im new, but trying real hard! I just need general direction, not details yet..i'll try to figure those...just looking to avoid brick walls...bottlenecks...inefficiencies etc upfront. Hardware: motorola vt2442 - trixbox Apps: Dot Net application that operates the Manager API and the FASTAGI interfaces. I have the 2442 set as a PLAR so as soon as the ext is off-hook, it dials into the *61 dial-plan. This is what I want to happen: Ext goes off-hook vt2442 PLAR auto-dials *61 *61 dial-plan gets some info from the inside user Need help for everything below: inside user is put on hold..queued or parked or put in a conf??? Sep call is originated to an outside line If call answered some info is gathered from the outside party outside party is joined to inside party. I am willing to use any or all of a combo of the following: Manager API FASTAGI dial-plans .call files or anything else!! The Manager/FastAGI application can perform any action required. The way that I can think of would be to create a context or add an extension in your outward context to call an AGI that would solicit the info you need and then push you into a conference. Then issue a .call file to call the other party, gather your info and then pop them into the same conference. Store the data gathered by each caller in the AstDB or another db like FirebirdSQL or MySQL. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db You indicated that your .net classes have a Manager implementation. Creating a call should be fairly easy using the Manager API originate: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate I wrote a built-in object into AsterPas to do call files which is just an abstraction of the System Command implemented in a FastAGI server: http://www.datatrakpos.com/pos/datatalk/asterpas.aspx There are some script/code samples on the site that may be useful in implementing that kind of thing through whichever .net abstraction layer you are using. Pascal is not that different from C#... Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help installing on OpenSuSE 10.2
make[1]: g++: Command not found You have just to install cpp Alex, Malcom Kemp a écrit : Thanks to all that have helped me so far. I have made a lot of progress. I am able to make prilib and zaptel. Now to Asterisk… After installing the kernel source, I have: # cd /usr/src/linux # make cloneconfig # make prepare-all Then I have run ./configure in the asterisk-1.4.4 directory. I have: # make clean # make Which goes through a number of compiles and then ends up with this: asterisk2:/usr/src/asterisk-1.4.4 # make menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [LD] stereorize.o frame.o - stereorize make[1]: g++: Command not found make[1]: *** [stereorize] Error 127 make: *** [utils] Error 2 Any suggestions would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra MWI
Anthony Francis wrote: Lee Jenkins wrote: Stephen Bosch wrote: Lee Jenkins wrote: I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Did you set the mailbox= variable in sip.conf? I made that mistake yesterday and wasted 90 minutes hunting for the problem. Hey guys, Thanks for responding. Yes, I have set that. Here is my sip.conf entry for that extension: [117] context=Management type=friend canreinvite=no dtmfmode=rfc2833 callerid=Napoleon Hill302-539- nat=no port=5060 qualify=no secrete=117 host=192.168.1.117 [EMAIL PROTECTED] disallow=all allow-g729 allow=ulaw is this actually autheticating? secrete=117 should be secret=117 Sorry. That was a typo from editing the copy/pasted data before posting it. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra MWI
shadowym wrote: This is probably cold comfort but I have NEVER had any issues with MWI working on Aastra phones. It always just works by default. No extra configuration necessary on the phone for sure. Just reset it to factory defaults. Explicit MWI is NOT checked by default and I have never had to check it. Yeah, that is what another responder said. I have MWI working on my polycoms, but can't seem to get it to work on Aastra's for some reason. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail Access
On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote:
Mike Hammett wrote:
I was looking at the ILECs? web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during the greeting to
enter the voicemail system? If they call their own number, how do I get
Asterisk to recognize that and take them to the voicemail system?
A common approach is to use the caller id in combination with some digit
sequence. For my systems, I've just used 555 as the VM extension.
exten=555,1,VoicemailMain(${CALLERID(num)})
For access to the VM from outside the system, I've used an AGI script to
query a database to validate the user.
It's also quite easy to set-up if you call your own extension number
from your extension it goes into voicemail for you extension.
You can have another number as above to access voicemail from another
extension.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
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Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
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[asterisk-users] Voice mail issue
hi, I am trying to check my voice mail on a new asterisk instalation. I get the standard voicemail menu, but when I press any button , it does not accept the option. It keeps repeating the menu and then exits ... Any suggestions ? thanks and regards Anand -- --- Trust in God , But Lock your Car --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoiceMail Access
If it is easy, could you enlighten me? I have another thread on caller ID
matching, but I haven't received any positive responses.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: Monday, May 21, 2007 5:55 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] VoiceMail Access
On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote:
Mike Hammett wrote:
I was looking at the ILECs' web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during the greeting to
enter the voicemail system? If they call their own number, how do I get
Asterisk to recognize that and take them to the voicemail system?
A common approach is to use the caller id in combination with some digit
sequence. For my systems, I've just used 555 as the VM extension.
exten=555,1,VoicemailMain(${CALLERID(num)})
For access to the VM from outside the system, I've used an AGI script to
query a database to validate the user.
It's also quite easy to set-up if you call your own extension number
from your extension it goes into voicemail for you extension.
You can have another number as above to access voicemail from another
extension.
Steve
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Re: [asterisk-users] Voice mail issue
Sounds like wrong type of dtmf signaling if you are using a SIP phone. Check the settings fordtmfmode in sip.conf. dtmfmode= rfc2833 ; Choices are inband, rfc2833, or info your choice might not have been right ;-) If your phones uses ZAP try relaxdtmf=yes Frank Gorgas-Waller Explido Software USA Inc. Phone +1-863-248-1195Fax +1-863-248-1155 EMail [EMAIL PROTECTED]ICQ 7733546 --QQ- We teach penguin to fly http://www.explido.us hi, I am trying to check my voice mail on a new asterisk instalation. I get the standard voicemail menu, but when I press any button , it does not accept the option. It keeps repeating the menu and then exits ... Any suggestions ? thanks and regards Anand ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call to an arbitrary outbound number by asterisk
If I had to make a wild guess, I'd expect that when you make a call off-campus you must dial an access code first. Looking at columbia.edu, I see that you're expected to dial '93' for a local number. 1+ is a number in the Centrex dial plan for the Morningside campus. http://www.columbia.edu/acis/telecom/tutorial.html#dialing Asterisk is dialing all 10 digits, just as you expected it to. Your dial plan is not prepending the '93,' so the campus pbx/centrex/switch/whatever thinks you're dialing an on-campus number and uses just the first five digits to complete the call. As '917' is a local call, is the '1' required? --Don Don Kelly CT Magic _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arpit Mehta Sent: Friday, May 18, 2007 1:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call to an arbitrary outbound number by asterisk Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the voicemail is running). I cannot understand why asterisk is doing this whereas my dialplan says it needs to connect to other number exten = _.,1,Dial(Zap/g1/19173995791) Also any idea if this is an Asterisk problem or a telco problem. Any help/hints/suggestions would be most welcome Here are my files. zapata.conf context=incoming switchtype=national signalling=pri_cpe group=1 channel=1-23 extension.conf [incoming] exten = _.,1,Dial(Zap/g1/19173995791) # I have added this line in the dialplan is because I want it to match the last 5 digit and simply dial the number 19173995791 such that a call leg is established between the calling party and the number 19173995791 CLI debug information -- Requested transfer capability: 0x00 - SPEECH -- Called g1/19173995791 -- Zap/1-1 is proceeding passing it to Zap/23-1 -- Zap/1-1 is making progress passing it to Zap/23-1 ### The call keeps ringing for sometime then it goes to voicemail. The message comes when the voicemail start. Note that I have not setup any voice mail -- Zap/1-1 answered Zap/23-1 ### Goes to the voicemail -- Native bridging Zap/23-1 and Zap/1-1 -- Channel 0/23, span 1 got hangup request -- Hungup 'Zap/1-1' == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
