Re: [asterisk-users] regcontext regexten
/etc/asterisk/extensions.conf /etc/asterisk/extensions.ael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of harry R Sent: Monday, August 10, 2009 1:22 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] regcontext regexten Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? Sure... let's say I have a phone with the following configuration in sip.conf: [myphone] type=friend context=inside host=dynamic ; phone will register w/ Asterisk secret=mysecret regcontext=some-context regexten=6123 Thank Jared. So I have one more and last question about regcontext. Where do asterisk create context some-context ? I see context by taping dialplan show some-context in CLI but I dont know in which config file it's created. Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
On an entirely unrelated note, do you have the gsm asterisk sounds installed? Maybe that vm-*.slin files dont exist. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Maciejewski Sent: Friday, May 22, 2009 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe not working with GSM codec? Hi Dhaval, The reason confno '12' is not found in meetme.conf is because I am using MySQL as realtime config backend. See few lines below there is: [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12' My meetme.conf: [general] audiobuffers=32 logmembercount=yes schedule=no 2009/5/22 DHAVAL INDRODIYA dhaval.it01...@gmail.com: can you look on this from your debug app_meetme.c:3030 find_conf: The requested confno is '12'? == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf == Found [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference its on line number 318 it seems that you doesent specify valid conference number can you post meetme.conf regards Dhaval On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Early Media
am i right in understanding that this feature is called color ring back tone? On Wed, Mar 25, 2009 at 8:16 PM, Danny Nicholas da...@debsinc.com wrote: Change line 2 to this: exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|m) this will play moh for 300 seconds or until the other end answers. The only issue you may have is that some carriers don’t generate a proper response when “answering” to the music would continue over your conversation. (ATT conferences in particular). -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Khaled W. Chehab *Sent:* Wednesday, March 25, 2009 9:36 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Early Media What I am meaning is . I want to start a music on hold and dial the number (009713045212) In the same time and when the call is connected the music will stop and I will talk to the called number Exten = 444,1,-- exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|) is it feasible regards *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, March 25, 2009 3:34 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Early Media YMMV, but you might try this Exten = s,1,background(background_song) Exten = s,n,Answer() ;start billing -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Khaled W. Chehab *Sent:* Wednesday, March 25, 2009 8:27 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Early Media Dears, - Anyone know how to play an early media as (background song) with no billing and when the call is connected the song will stop and the billing starts. Regards -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $20 Bounty
I wish my family and I could live on $40 a week... simplify, simplify, simplify -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
Ideally the person needs to enter the credit card number, expiration date in mmyy format (which is the format in which the expiration date is shown on the card), and the ccv number. The amount would probably be calculated on the basis of the outstanding amounts, or the products selected. Think of trying to buy a plane ticket or pay a bill. You are unlikely to want the caller to enter the amount. The thing is to structure the IVR in such a way that the caller is informed of the amount and does not have to enter it. If you take a far out case of a donation help line, you can simply go for $5 press 1, for $10 press 2, for $20 press 3. If someone wanted to donate $15, too bad for us. If it turns out a lot of people want to donate $15, you can simply adjust the IVR (and of course the other logic). This is a simple enough task. The big deal is supposed to be in ensuring that the date and the ccv number DTMF do not show up in any log files or trace files, and surely do not get logged by the application. You can simply turn off all DTMF logging, but you dont want to do that. Only the place where you accept the secure information, the logging should be absolutely turned off. Getting the issue? On Wed, Feb 18, 2009 at 11:20 PM, bilal ghayyad bilmar...@yahoo.com wrote: And is there a bank accept to give such kind of communication? The user was able to dial his card number and the amount from his phone (or IP Phone registered with Asterisk), and Asterisk communicate with the bank or company credit card provider? How the user will enter $50.25? What about expiration date of the credit card? Regards Bilal Our creditcard company's small print _insists_ on a direct analog exchange line with no other devices in between. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk You can do it an interface using AGI to comunicate with equipment or verifone. I did it once ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can anybody tell me how Magic jack can be so cheap ????
the ads will start once there is critical mass. the following are the scenarios for ads: 1. when you dial a number, before hearing the ringing, you have to listen to an ad. the length of the ad would be proportional to the intensity of your usage... the more you use, the longer the ads. 2. when the caller answers, they will first hear magic jack promo, then they will hear your voice. 3. the call in interrupted every few minutes to play an ad to both parties. 4. they will give an ad free service if you pay a higher charge. I just hope I am not giving the people at magicjack any ideas, but if I am, I would sure appreciate if they pay me!! On Sun, Feb 8, 2009 at 9:58 AM, k4...@bellsouth.net wrote: Never seen any ads except for them. Actually the thing sits on a server down in the garage so I don't see anything! Darn thing just works! I bought it as a second line when the wife is using the copper line to work. Ronny K4RJJ -- Original message from Forrest W Christian f...@mt.net: -- Or more accurately, they believe they can follow the NetZero or Juno model (Free in exchange for ads being pushed to you). -forrest C F wrote: They believe they have advertisement revenues. On Sat, Feb 7, 2009 at 5:45 PM, Ignacio Ortega A. wrote: How Magic Jack can only charge $20 per year? do they have a call limit? do they have a call duration limit or limit of minutes per day?, Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Autodialler query
Sriram: whats going on here?? unless you are developing a vas, in which case, the provider for whom you are doing this will have to help you. each provider would be doing this differently. regards Kinjal Dixit On Thu, Feb 5, 2009 at 7:20 AM, da...@iaxtalk.com wrote: Hi Sriram, the customer should be billed a premium rate ex, Rs.9 per minute.. Will be billed by you or by telecomm company? Regards David - Original Message - *From:* Sriram d_r_sri...@hotmail.com *To:* asterisk-users@lists.digium.com *Sent:* Thursday, February 05, 2009 1:46 PM *Subject:* [asterisk-users] Autodialler query Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents who talk on certain subject - but the challenge here is that the moment he presses 1 - the customer should be billed a premium rate ex, Rs.9 per minute.. Is that possible ? If yes then can anyone guide me as to what all points i need to focus on during my discussion with operator ? Thanks Sriram -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted information
Hyelo Ambarish: http://www.asteriskwin32.com/ Go for it. Kinjal Dixit On Thu, Jan 29, 2009 at 4:17 PM, ambarish.deshm...@wipro.com wrote: Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with AudioCodes Mediant 2000
hi I am trying to get Asterisk to work with Mediant 2000. I have searched and found a few articles on the topic, but none of them seem to solve my problem. My knowledge is weaker on the gateway side. To begin with, I think I should get Softphone to dial through the gateway. I need to know how to configure the Mediant 2000 so it can act as a SIP proxy. The asterisk side would be easy enough to figure out. 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
