[Asterisk-Users] Re: [Asterisk-Dev] Getting info about changes in CVS
Eric Wieling wrote: There are several ways to know what changes in Asterisk's CVS. This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly up to date CVS changelog summary information. You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs Archives of the Asterisk-CVS mailing list are at http://lists.digium.com/pipermail/asterisk-cvs/ Is there any reason that CVSView is not installed and publically viewable? It might help the who don't know all the CVS CLI commands get a graphical (and colored) view of the lines added and changed over time? Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager Interface "Action: Originate" changed
Tony Wasson wrote: I have recently noticed that the "Action: Originate" options in asterisk 1.0 CVS has changed sometime between 2/23 and 3/18. To post a follow up for posterity, 2 tips were suggested when using the Manager Interface: 1) Make sure to supply Context AND Priority when using an Exten. NOTE: It used to work without a Priority, but not anymore. 2) While not a "hard and fast" rule, capitalization may help. Action: Originate Exten: 200 Context: stations Channel: SIP/agent007 Priority: 1 Here's a brief troubleshooting checklist if Manager Actions like Originate are failing: 1) Make sure Asterisk is starting with "debug mode" by starting it with a few -vvv's after it. I'm using this in my /etc/inittab ax:2345:respawn:/usr/sbin/asterisk -vvvcf You can also just stop asterisk (asterisk -rx "stop now"), and relaunch it like this: # asterisk -vvvcf 2) While making the call, monitor the console for any errors using # asterisk -r 3) Ensure any dependant devices are actually connected and registered... "sip show peers" "iax2 show peers" While getting the rather unhelpful message of: Response: Error Message: Originate failed I got this messages on my console thanks to debugs: Mar 31 10:03:24 NOTICE[21526]: app_dial.c:536 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy at this time After some investigation of "sip show peers" I diagnosed my problem as an SIP device (Audiocodes MP-108) that needed to be rebooted. Hope this helps you out! Tony Wasson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager Interface "Action: Originate" changed
I have recently noticed that the "Action: Originate" options in asterisk 1.0 CVS has changed sometime between 2/23 and 3/18. I have a 2/23/04 CVS installation (cvs checkout -r v1-0_stable asterisk ) that allows me to make calls like this using the Manager Interface on port 5038. action: login login: admin secret: mypass action: originate exten: 200 context: stations channel: SIP/agent007 I have a 3/18/04 CVS installation that does NOT work the same way. Entering the same information in spits out Response: Error Message: Originate with 'Exten' requires 'Context' and 'Priority' So I've tried adding a priority of 1, like this: action: originate exten: 200 context: stations channel: SIP/agent007 priority: 1 I simply get: Response: Error Message: Originate failed Obviously, something in the Manager code has changed. With the newer code I am unable to originate calls. Can anyone shed additional light on how to originate calls under the new 1.0 style Manager Interface? Tony Wasson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two-stage dialing
[EMAIL PROTECTED] wrote: I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) You can do "2 stage dialing" with a call file. This may be a hack, but it currently works. I feel it is backwards from all the samples I saw since it rings the Agent's SIP phone first. From what I see, call files connect the Channel first, then run the Context/Exten/Priority after it's picked up. #Two stage dial call file #Rings the Channel first the connects the "Extension". #agent's SIP phone, like SIP/agent007 Channel: SIP/xws108 #Use a context that can dialout. #Probably whatever the agent's phone is set as Context: internal #Phone number to ringYou may need a '9' prefix #To Asterisk, Extensions don't have to be internal only... Extension: 5551212 #use '1' unless you know what you're doing Priority: 1 #If your telephone interface sends CallerID, be sure to set it. Callerid: 888-555-1212 - Make this file and then copy it into /var/spool/asterisk/outgoing. Don't try to make the files in that directory, because Asterisk may read it before your script is done writing the file. This may make Asterisk mad. P.S. You can also do this with the Manager Interface, I lifted enough perl code to make this work for me using a "Click to Dial" script that sucks in the phone number from a web interface. http://www.azxws.com/asterisk Tony Wasson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Dynamic Extensions
Darren Nay wrote: Hello All, I am just looking into Asterisk as a viable voicemail solution for our phone service. In order to use it though I will need to make extensions.conf dynamic (ie. Via MySQL). Is this possible? Sure.. Set up the database as you see documented. You can schedule this up to once a minutes using crontab. Setting something like this in your /etc/crontab should do it nicely */5 * * * * root /usr/local/sbin/update-voicemail 2>&1 > /var/log/vm.log NOTE: You can increase the frequency by using */3 or */1. */5 means every 5 minutes. */3 means every 3 minutes. Then make /usr/local/sbin/update-voicemail look like: #!/bin/bash /path/to/retrieve_extensions_from_mysql.pl /usr/sbin/asterisk -rx "extensions reload" Next, make the script executable #chmod +x /usr/local/sbin/update-voicemail This is not totally dynamic, but it ought to be close enough. You could make this completely dynamic using a trigger. Tony Wasson I've found the following information on this subject: http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql <http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql> However, this is not a fully dynamic function. It requires me to pull the mysql database every so often (presumably via cron) and then restart asterisk after updating extensions.conf. Is it possible to setup asterisks so that extensions.conf is fully dynamic via a MySQL database? Thanks for the help!! Regards, Darren Nay [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "Click to Call" Perl CGI script - TACI
I've hacked up a little "click to call" web CGI in Perl using the Asterisk Manager Interface and Net::Telnet. I've called it TACI - Trivial Asterisk Call-generator Interface. http://www.azxws.com/asterisk/ This is far from perfect or finished, but it should be a start for anyone looking to make "clickable" URLs that spawn phone calls. It has a rudementary HTML interface to make a call. You can also make calls in a URL (providing a real click to call). Finally, you can use it from the command line for debugging and other purposes. You'll need to specify a context, SIP account, and extension to call. This seems to work well with the 100 or so calls I've tested. This should be easy to extend for anyone using IAX also.(It is hardcoded for SIP on 1 line). The README should get anyone interested in doing this going. Hope this of use to you! Tony Wasson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
