Re: [asterisk-users] Best VoIP conferencing phone ?
We've been happy with the polycom IP 7000. Darren Wiebe On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote: Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue *1)Polycom SoundStation IP 7000 * *Why it's best: *The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. * * *2) Polycom Voicestation 500* * * *Why it's a best pick: *The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. * * *3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S* * * *Why it's a best pick: *With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. * * *4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone* * * *Why it's a best pick: *The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. * * *5)Polycom SoundStation VTX 1000* * * *Why it's a best pick: *The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000* 7) GXP2120 6-line Executive HD IP Phone* On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. ** ** Regards, ** ** Faisal Hanif ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 30, 2011 11:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Best VoIP conferencing phone ? ** ** Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
Why not firewall hack attempts after 3 tries? When we started doing that the quantity of hacking attempts dropped right off. We also setup our own fail2ban sharing server so that we could share the bans across multiple servers. Have a look at http://www.f2bshare.org/index.php?title=Main_Page if you want to do something similar. Why try to make Asterisk into something it's not intended to be? Just use your firewall for what it's good at. -- Darren Wiebe On 7/23/11 11:38 AM, CDR wrote: I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user information, and also, b) disable any response to any REGISTER packet altogether. Can somebody please write patch? Or should we go broke trying to stop the flood of criminals coming from abroad? Federico On Sat, Jul 23, 2011 at 1:00 PM, asterisk-users-requ...@lists.digium.com wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: use dahdi for local terminal modem access? (Lyle Giese) 2. dialplan pattern help (Armand Fumal) 3. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined (Patrick Lists) 4. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined (Paul Belanger) -- Message: 1 Date: Sat, 23 Jul 2011 09:29:26 -0500 From: Lyle Giesel...@lcrcomputer.net Subject: Re: [asterisk-users] use dahdi for local terminal modem access? To: asterisk-users@lists.digium.com Message-ID:4e2adac6.4010...@lcrcomputer.net Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 07/22/11 22:47, William Stillwell wrote: Um, no VOIP involved here. Wrong. What do you think Asterisk is? Chopped meat? It's a VoIP switch. All traffic inside Asterisk is VoIP. I have an asterisk server with 2 23B+D PRI's I want to telnet/ssh into the asterisk server, and make an outbound call serial based modem/terminal connection (Like the 80/90's BBS Days). No TCP/IP or PPP or crazyness (ie, dialing into a Modem set to AA hooked to a Cisco Console Port) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Friday, July 22, 2011 8:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use dahdi for local terminal modem access? On 07/22/11 18:13, William Stillwell wrote: I have some terminals that have phone lines. One of my tech had an idea of using IAXmodem or something similar to use existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console. Anybody ever heard of doing this? I would think maybe would use iaxmodem maybe and a shell terminal app? (basically I'm dialing into a remote access device that uses a pots like for remote administration, and don't want to string a channel bank off my asterisk box, and a hook to a modem) -- Depends on your expectation. Because of compression in the codecs, it will be hard to get fast dialup. If you mean ssh or telnet, it might work. If you mean vnc or RDP over this, you may not get enough usable bandwidth to do that. Given this, I have in an emergency dialed into a RAS server via a VoIP line. My laptop connected at 14,400bps. All I needed to do was telnet into an APC masterswitch to toggle power on one outlet. It worked. I was surprised at getting a 14,400bps connect. I was not expecting that high and really did not need that high. 300 baud probably would have been fast enough to telnet into an APC masterswitch. Lyle Giese LCR Computer Services, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory
[asterisk-users] Sharing Fail2ban data
Good Day, I've been doing a little work that I wanted to share. We've had a number of Asterisk systems that have been under heavier than normal attack. We use fail2ban but we either have to let each system be exposed or keep all the data synchronized which is a bit of a chore. I wrote a little server that assists in keeping data synchronized across sites. If you're interested in using it to assist in managing your own fail2ban sharing list I'll gladly share it. I also am offering it as a free service for those who are interested in contributing to a blacklist. If you're interested the information is available here: http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in the server code just drop me an email. Darren Wiebe dar...@aleph-com.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to switch on electric heaters remotely?
We recently completed a project using products from here: http://www.controlbyweb.com/webrelay/ They were easy to setup and can be controlled in a variety of fashions included http queries. Darren Wiebe On 18/10/2010 8:34 AM, Marco Signorini wrote: Hi Did you looked at Arduino + Ethernet Shield? Is something you can program in C or C++ to receive a simple TCP and/or HTTP packet and turn on an external relay. From the dialplan you can run an http query through curl and/or an external AGI command. Best regards, Marco Signorini. -- Marco Signorini http://www.ethermania.com http://www.ingegnitech.com Roberto Piola wrote: we're using a Damocles Mini (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of course, the damocles will have to drive a high-power relay. the damocles can be driven via snmp, so you'll have to simply call the snmpset unix standard utility On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades list-aster...@skycomuk.com wrote: Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only control 750W so you will probably need to get it to control a more powerfull relay as a heater is going to take a lot of current. It can be controlled by a virtual serial port so you just program the extension to make a system() call to a simple script which sends a string of characters to the serial port. That device is quite expensive. You could probably find something much cheaper on ebay. Gilles wrote: Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over running my did's
On 10/04/2010 9:24 PM, Timothy C Litwiller wrote: I have a did with 20 channels from didforsale. that we use to let local members call to listen to a conference several times a week without long distance charges. The upcoming call is getting more interest than usual and from places that are not local so we want to use a free conference service in addition to the local conference. How can I setup a conference on my asterisk box for the people that normally call in there and also call an outside number for those that are above and beyond the 20 lines channels I can provide and the are long distance anyways so a number here or a number in iowa doesn't make them any difference. is there a way that I could call the outside conference # and then transfer it to a local asterisk conference and then hang up can call the local asterisk conference back - and if I do that how do I hang up the long distance conference when it is done? I seem to be missing some basic understanding here. I would call into the free conference service and then transfer that call into my meetme conference. If you're using Trixbox you can use the MeetMe web control to disconnect the call when you're done. You can also disconnect calls from the asterisk cli using the soft hangup command. Darren Wiebe dar...@aleph-com.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
Steve Totaro wrote: On Tue, Oct 13, 2009 at 2:41 PM, SIP s...@arcdiv.com mailto:s...@arcdiv.com wrote: David Wathen wrote: Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. Thanks, David Wathen Depends on what level of firewall you're looking for. For a full firewall on either a dedicated system or one of your own, I cannot strongly enough recommend Astaro Linux firewall. Better throughput than a pix, worlds easier to operate and configure, and comparable in price. Very SIP/VoIP friendly. Loads of optional modules (we use its mail filter module to filter spam/viruses for several hundred thousand user mailboxes, for instance) to limit the cost to what you need. Also has a built in SIP Proxy, although I've never used it. Excellent platform. Of course, at home, I just use a little Linksys WRT box. It's hardly a corporate-grade firewall, but it's quite SIP-friendly. N. No votes for Vyatta? I have been seriously checking it out. Thanks, Steve T I played with a demo of Vyatta and it looks pretty good. We've been using mostly Endian (www.endian.com) or M0n0wall. I've had good luck with both of those. Darren Wiebe dar...@aleph-com.net ___ -- 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] Messaging System
Ricardo Melendez wrote: Hi to All, I need to implement an automatic telephone messaging system that works like this: -the system generates the call based on mysql records or any database -when the client answer the phone, the Asterisk PBX playback a recorded message -when finish, hang up the channel. Only for voice messages not SMS. Exists some application based on Asterisk that makes this, or any code to implement in dialplan Thanks in advance. Ricardo ___ -- 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 We've released an application to do that on www.callblast.org. -- Darren Wiebe dar...@aleph-com.net Aleph Communications www.aleph-com.net ___ -- 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] Zopier Client
Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg I've been using it on my notebook. I've been happy with it but I'm not a heavy user. The biggest reason I purchased a few copies of it is that I need to have several different sip and iax2 connections for testing purposes. -- Darren Wiebe dar...@aleph-com.net Aleph Communications www.aleph-com.net ___ -- 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] Inexpensive device for bandwidth management
My thoughts were similar. Availability has not been a problem for us on the WRT54GL boxes. We're pulling them out of our wholesaler all the time without any problems. Darren Wiebe dar...@aleph-com.net Jeff LaCoursiere wrote: And why not DD-WRT, which runs on many more platforms including some more recent platforms still selling on shelves? :) j On Sun, 5 Apr 2009, Mike wrote: I just reread my question and realized I might not have been clear enough. What I meant is that it only seems to works on older Linksys hardware revisions. How do I make sure I can get those? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Sunday, April 05, 2009 15:30 To: oliv...@hh174.be; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Inexpensive device for bandwidth management Actually that was my original thought. BUT?according to what I read on their FAQ, the hardware that can be used is rather limited. How do I secure a reliable supply of those? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hh174 Sent: Sunday, April 05, 2009 14:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inexpensive device for bandwidth management Linksys (cisco)WRT54GL and the tomato firmware. 5 minutes setup Olivier Mike a ?crit : Thanksthe thing is I need many device (one for each of my hosted customers) and I'd like this process to be as easy for non-techies as possible, because some of those are technologically-challenged, and need to install the box by themselves or with the help of an IT person that only knows how to install a run of the mill router. So an out-of-the-box thing would be better, but I was recommende the pfsense before and will take a look at it. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Sunday, April 05, 2009 13:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inexpensive device for bandwidth management The following two links deal with the same familly of boxes. Generally it's $20 for a case, $20 for a powersupply, but you've probably got an old one that will work. and almost all of their boards are under $200, except for the ones with lots of gigabit interfaces. Many are under $100. http://www.mikrotik.com/ http://routerboard.com/ On Sun, Apr 5, 2009 at 11:07 AM, Mike mailto:l...@virtutel.ca l...@virtutel.ca wrote: Hi, I'm looking for a good network device that does bandwidth management. It can be integrated in a router or stand-alone, but must be SIP-friendly. I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest hardware revisions can't downgrade to the version that worked well) and the DI-724GU (SIP-friendly, but bandwidth management is automated and not configurable enough for my taste), both from D-link. What else is out there and allows me to do upstream QoS on cable/DSL links? Both D-Link routers were under 200$ (99$ and 159$ respectively) and were perfect price-wise for my target customers. Mike ___ -- 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 -- Drew Einhorn ___ -- 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] HDD FULLL
Just restarting it won't do anything. You could use the following command to find any files over 200mb on the system. Be careful about blindly deleting stuff though *find / -type f -size +200M Darren Wiebe dar...@aleph-com.net * David @ULC wrote: I have 320 GB SATA HDD. When I checked my phpsysinfo, it shows 95% HDD is filled. [r...@vicidialnow mailto:r...@vicidialnow ~]# df Filesystem 1K-blocks Used Available Use% Mounted on /dev/sda2 301924504 285002780 1337472 100% / /dev/sda1 101086 11062 84805 12% /boot tmpfs 1553832 0 1553832 0% /dev/shm [r...@vicidialnow mailto:r...@vicidialnow ~]# du 16896 . You have new mail in /var/spool/mail/root [r...@vicidialnow mailto:r...@vicidialnow ~]# df -i Filesystem Inodes IUsed IFree IUse% Mounted on /dev/sda2 77922304 528483 77393821 1% / /dev/sda1 26104 34 26070 1% /boot tmpfs 219910 1 219909 1% /dev/shm You have new mail in /var/spool/mail/root But my concern is how to solve it I even tried restarting the server , though it will kill unwanted process and will release the space but no ho ___ -- 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] Please help test the gender detection module at 575-613-4392
Asterisk Asterisk wrote: You have some good points. Justin Newman isn't exactly someone we don't know. However I only I agree that my name wasn't clear, but I was trying to avoid getting a bunch of spam myself. I'm not sure if I've personally ever spammed the list and I'm pretty supportive of the community. I have been part of these lists for many many years. * The message starts by asking you to call a number. That was the help needed and it worked. There have been more than 500 different callers now and they keep coming in. I'm going to need help with a second round of testing, after I release the updates today and Sunday, but I haven't figured out how to entice people to test again. I thought about doing an outbound call and most people probably wouldn't care, but I'm anti-spam myself and that sounds like spam to me! Any thoughts? -- Snipped -- I'll be happy to try it again to see if I've become a male yet. :) Darren Wiebe dar...@aleph-com.net ___ -- 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] Please help test the gender detection module at 575-613-4392
Pretty cool. I'm almost offended though as I'm not usually guessed as a female of the species. :) Darren Wiebe dar...@aleph-com.net Asterisk Asterisk wrote: Steve, Tried to test and got call could not be completed as dialed. Were you able to connect? If not, please try again. Call volume has been growing. How about a moving stress variable that could be used as a lie detector of sorts or just to measure how certain parts of a script, or certain questions may This is possible. Do you want to call or e-mail to discuss? I guess to get a baseline, you would have to ask a few inert questions. Yes, I definitely need to do this and will probably add this in for the next release. Justin Newman nt_jnewman at yahoo.com *From:* Steve Totaro stot...@totarotechnologies.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Wednesday, February 18, 2009 10:57:47 AM *Subject:* Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro stot...@totarotechnologies.com mailto:stot...@totarotechnologies.com wrote: On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk nt_aster...@yahoo.com mailto:nt_aster...@yahoo.com wrote: This module detects gender and approximate age range. I'm working on getting it's accuracy to 80%+ on a consistent basis, after implementing filters to remove background noise and other artifacts. It's designed for a number of things. To start, I have several clients (primarily mobile content and servers providers) that want to profile and generate demographics of their users for selling advertising. They also want to understand their user base. Plus, some customers have found that male and female users tend to respond differently to different prompts, flows, etc. This helps in designing a system that meets needs of many different types of users. Of course, there are many other uses and I'm sure people can generate some cool ideas. Let me know how it works when you try the test number at 575-613-4392. Also, let me know if you have any interest in the module. Justin nt_jnewman at yahoo.com http://yahoo.com *From:* Ron Joffe ron.jo...@gmail.com mailto:ron.jo...@gmail.com *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Cc:* Asterisk Asterisk nt_aster...@yahoo.com mailto:nt_aster...@yahoo.com *Sent:* Monday, February 16, 2009 11:05:24 AM *Subject:* Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 That's an interesting module. Care to elaborate on what you designed it for ? Thanks, Ron On Monday 16 February 2009 13:29, Asterisk Asterisk wrote: I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others call too. Also, let me know if you have any need for the module. Justin Newman nt_jnewman at yahoo.com http://yahoo.com Tried to test and got call could not be completed as dialed. This sounds very interesting Justin. -- Thanks, Steve Totaro Justin, how about building some additional functionality. How about a moving stress variable that could be used as a lie detector of sorts or just to measure how certain parts of a script, or certain questions may prove to be more stressful where simply rewording them may have a less stressful response? I guess to get a baseline, you would have to ask a few inert questions. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 -- Darren Wiebe dar...@aleph-com.net Aleph Communications www.aleph-com.net ___ -- Bandwidth and Colocation
Re: [asterisk-users] Looking for SIP loud ringer
We've done this with good results. You can also get one that flashes a bright light for not a lot of money. Darren Wiebe dar...@aleph-com.net Steve Gladden wrote: If you wanna go low tech. down dirty you could also go with a conventional POTS phone line 'loud ringer' device and simply hook it to an ata such as a PAP2, and add the PAP2 to the ring group. Why don't you put a PC in the storeroom with a softphone to be the loud ringer? You could make the ring though the speakers be as loud as the system would support. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, January 28, 2009 9:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Looking for SIP loud ringer Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Current Open Source Billing Package
Jerry Jones wrote: After spending a couple hours scanning for an open source (non- commercial) billing package yesterday I am underwhelmed. Almost all of the packages listed on the WIKI appear to be defunct, for several years now. I will be happy to get a login and edit them out if that is the proper method to do so. My requirements are very minimal and at this point unless I have missed something will just write my own. I do not do calling cards. I have no near term need for the package to actually talk with asterisk at all, other than to import the CDR either via files or as a login to MySQL. I do have monthly recurring charges which need to be included monthly. I do occasionally have need to one off (manual) billing charges. Rating for calls would be nice but not mandatory ( we have very minimal International). Ability to export to an accounting package a plus. Ability to generate hard copy Invoices and/or email them to the cust. Ability to generate a list of current Invoices. Runs on Linux. All in all not a very complex set of requirements, but the few packages that seem to be currently offered generally do not fit the bill. Yes there are many commercial packages, but unless they are very minimal in cost I have no interest in them. So my question is, have a missed a golden nugget out there? tia Jerry Have a look at astpp (www.astpp.org) along with OSCommerce. This should do what you're looking for and you do not need to link to Asterisk, etc. Darren Wiebe [EMAIL PROTECTED] ___ -- 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] implementing an intercom with asterisk
For simple paging the bogen tamb works very well. Just hook it up to an fxs port and you're good to go. Darren Wiebe [EMAIL PROTECTED] Jonathan Disher wrote: I am looking to replace the phone system at my father's shop with an Asterisk box and some Cisco phones, but one piece of the implementation is tripping me up. He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Currently with the ancient Meridian system installed, there is a paging intercom (to page employees, etc) on a dedicated extension - play a loud tone, then set up a 2 way channel. Anyone got any ideas, hardware wise, on how I might implement this with an Asterisk system? Thanks, and if this isn't appropriate for this list, if anyone has a better destination for the question, Id be quite appreciative. -j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Least Cost Routing
ASTPP (www.astpp.org) will do calling cards / prepaids as well as lcr. Darren Wiebe [EMAIL PROTECTED] emist wrote: Hello, does anyone know of a good calling card solution for asterisk that is able to do lcr? Does astcc do this? I've been searching around and I can find some lcr modules/apps but none that incorporate prepaid card functionality. Regards, Igor H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Realtime status feature - user feedback needed
Just FYI, I wrote an application that tracks the status of SIP or IAX2 extensions by listening to the AMI. It was for use by callshops but would probably require minimal change to work for you. It's currently part of the ASTPP source code. Darren Wiebe [EMAIL PROTECTED] Atis Lezdins wrote: On Thu, May 8, 2008 at 3:49 AM, Ex Vito [EMAIL PROTECTED] wrote: On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Tilghman Lesher a écrit : Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. True, that was one of initial options, however I prefer to NOT have yet another layer. I will consider this as an option where appropriate. However this looks quite awkward to me, somehow it reminds me tailing queue_log or CDR and putting result into MySQL database.. just one level more that way. For now, I see only one point against this - having status cleared upon module load/unload makes it easier to follow restarts/module loads. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. Agreed. There should be ONE to do it, it should be SIMPLE and as RELIABLE as possible, without interfereing (bad spelling?) with asterisk's operations: the proxy into AMI looks like the way to acheive the required funcionality... After all, that's exactly the purpose of AMI ! Let's keep the codebase as small as possible, let's make asterisk as solid and reliable as possible. Let's not reinvent wheels! Ok, so we're exactly at the point. Yes, I agree that it would act nearly the same way as AMI actions, however there's one great advantage - It would be really easy to set this up for user. AMI proxy would take more effort, need configuration, etc. Then there should be much more development support for proxy than for code within asterisk (if you have noticed, there's no new code, just reusing existing functionality) I think that there should be several ways how to do something, not just one. Having realtime status won't mean that much changes, for now I can see only 4 families for this - queue_members (already existing), queue_callers, channels and meetme. Really nothing more to give full overview of Asterisk Status. Regards, Atis ___ -- 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] Dialplan, Extensions, etc. Worksheet
If you're willing to cc me a copy I'll be in your debt. Thanks, Darren Wiebe [EMAIL PROTECTED] Steve Totaro wrote: On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Has anyone created a worksheet they can share for designing a dialplan, extensions, voicemail, etc. I'm making my way through the O'Reilly Book (dead tree version) and finding it enlightening. I have hacked at dialplans created by others but never actually came up with a design for my own system. It's sort of a work in progress made of bits and pieces from all over. Having a real plan would probably make things easier. Rod -- Rod, You will be glad that you are taking the learning curve plunge down the road. No pain, no gain. I can certainly say that I am glad I got into Asterisk way before there was any real documentation or GUIs for that matter. It forced me to learn the real deal Asterisk through trial and error which is invaluable if you plan on really getting into it. Then again, if you want easy, use a GUI. Easy isn't what I'm after. I was hoping for planning worksheets. Something to go over with a customer (I know I said this was for my personal system but that is the first step). How many extensions/ phones/ softphones, and what their /numeric/ extension will be. An IVR plan and the text that goes with it, voice-mail handling and mailboxes, etc. This type of stuff. So from the minimal number of responses -- yours :-) -- I'm going to guesstimate no one has anything like this at all or that they can or are able/willing to share. Out comes the notepad and the thinking cap. /-| Cheers, Rod -- Thanks, Steve Totaro Hey Rod, I think I may be able to help with worksheets from 3com, NEC, and other system vendor's sales channel. It obviously will not match exactly to Asterisk but will give you a great foundation for the functions and features that you need to question. I have my own but I prefer not to put it in the public domain. It is adapted from a conglomeration of many different proprietary systems that I have dealt with. I think many others have the same and consider it proprietary internal information for their business. Let me see what I can dig up from my archives. Thanks, Steve Totaro ___ -- 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] prepaid on the trunks
Am I correct in thinking that one application of this would be monitoring what you have left for funds with a prepaid vendor? Darren Wiebe [EMAIL PROTECTED] Brian J. Murrell wrote: On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Wouldn't you just equate a Calling Card (that's the unit that has an account balance and charges against it) with a trunk instead of a user or extension? You can call the astcc agi script with any value you want for a Calling Card identifier. b. ___ -- 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] prepaid on the trunks
Ok, I'm not aware of this feature in astcc and I can't speak for astbill or a2billing. I do know that I coded it into astpp and it's called vendor rating in there. It works but it's not used a lot at present. Darren Wiebe [EMAIL PROTECTED] Nhadie Ramos wrote: hi sir, yes that would be it, but instead of having a prepaid provider, i will setup my own as5300 and asterisk will talk to that. is that possible in astcc, astbill or a2billing? regards, nhadie Am I correct in thinking that one application of this would be monitoring what you have left for funds with a prepaid vendor? Darren Wiebe [EMAIL PROTECTED] Brian J. Murrell wrote: On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Wouldn't you just equate a Calling Card (that's the unit that has an account balance and charges against it) with a trunk instead of a user or extension? You can call the astcc agi script with any value you want for a Calling Card identifier. b. */Nhadie Ramos [EMAIL PROTECTED]/* wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Regards, Nhadie On Tue, 2008-04-22 at 22:59 -0700, Nhadie Ramos wrote: i want to create a billing system to monitor only the trunks and also to load amounts on those trunks. is this possible? will i be able to use app_prepaid for this? TBH, I don't really understand your description, but I will say that I implemented astcc a week or two ago and it works for what I need. Cheers, b. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ%20___ -- 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 Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ%20 ___ -- 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] AdvancedVoIP Billing ?
I'll jump in here. As the author of ASTPP, I have gone to considerable effort to make it so that ASTPP does NOT need to eat a whole system. All that you really need on the asterisk box to get ASTPP working in terms of asterisk requirements, is to make sure that the cdrs in the database have an accountcode set. You do not need to use it to manage your dids and extensions, etc. Darren Wiebe [EMAIL PROTECTED] Vicky wrote: I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, *Noc Phibee* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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] AdvancedVoIP Billing ?
Ok, cool. If you run into problems, please post at forums.astpp.org on the the astpp mailling list. Good luck, Darren Wiebe [EMAIL PROTECTED] Vicky wrote: I will definitely give it a try again to astpp then . I actually saw its online demo and was bit confused . I thought its managing extensions and all and i will have to start from scratch so i didnt gave it a try .It is a great software but only thing holding me back was thought that i will have to start from scratch :P . On 18/11/06, *Darren Wiebe* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'll jump in here. As the author of ASTPP, I have gone to considerable effort to make it so that ASTPP does NOT need to eat a whole system. All that you really need on the asterisk box to get ASTPP working in terms of asterisk requirements, is to make sure that the cdrs in the database have an accountcode set. You do not need to use it to manage your dids and extensions, etc. Darren Wiebe [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Vicky wrote: I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, *Noc Phibee* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://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 http://Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://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] How would you go about calling a list of numbers and 'speaking' a message?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have a script to do this found here: http://www.astpp.org/index.php?n=Misc.AutoDialOut Darren Wiebe [EMAIL PROTECTED] Tom Engleward wrote: --- Angus Comber [EMAIL PROTECTED] wrote: I have been asked by a client to process a list of telephone numbers. Asterisk should call each number in turn and if the recipient of the call answers, play a message - eg from a wav. How would I go about doing that? Make your message as /var/lib/asterisk/sounds/custom/mymessage.wav Then you'll need to create a context in extensions.conf like: [my-outgoing] exten = s,1,Playback(custom/mymessage); exten = s,2,Hangup then write a script to: 1. Read a single number from your list of numbers 2. Write that number into a .call file 3. Copy that .call file to /var/spool/asterisk/outgoing 4. Repeat for the next number in your list Asterisk will immediately make the call when the file shows up in the outgoing directory, unless the timestamp on the file is in the future. The .call files which you dynamically generate (one for each number) look something like this: Channel: IAX2/yourpstnprovider/numbertocall MaxRetries: 1 RetryTime: 5 WaitTime: 60 Callerid: whatever Context: my-outgoing Extension: s Priority: 1 WaitTime is how long to wait for an answer before giving up. See this page for details: http://www.voip-info.org/wiki-Asterisk+auto-dial+out __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEWUGC4DADnh+tnOQRAu9zAJ9Ac65YVXDaFYgV/TJB8DXtvKzOVgCfcLI0 gYK4juJx+BPnoYlFpZ4E2NM= =yudc -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] billing realtime
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jon, we can do that using ASTPP. The downside is that we don't currently have a way to limit the call lengths so that when they have multiple calls in progress they still can't go over their prepaid limit. On postpaid accounts this is not usually an issue but on prepaid it still is. Darren Wiebe [EMAIL PROTECTED] Jon Farmer wrote: JP Carballo wrote: Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFETxRg4DADnh+tnOQRAuhJAJ9kzGiQYh4Z6WPXXes6TKtwusBliwCeMvHG 3nrqsxdXNrfJbCZ3uzlpd5w= =+fV+ -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] Can Astcc allow dialing phone number more than once
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 chawki hammoud wrote: With a few fairly minor programming revisions to the script this would be possible. At present, ASTCC does not support that though. Darren Wiebe [EMAIL PROTECTED] Hi users: astcc script exits when dialing an uncomplete or wrong number. What changes need to be made for astcc.agi to allow dialing phone numbers more than one wrong attempt. Regards; Chawki Hammoud __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEQo+P4DADnh+tnOQRAtIhAJ9/VkKOG8IWQBN77arxiOVCerIOvgCdEkW5 tw8gdx/elcZZqM8ghJcqI7A= =baH/ -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] How to terminate ringing call before it is answered
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial Checkout options H and h. Darren Wiebe [EMAIL PROTECTED] Obelix wrote: Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because you current number it is not being answered, and you don't want to hangup before dialling again. /Obelix ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEPcn24DADnh+tnOQRAjzdAJ9ZOQQZ2OHXtZCT1kDiT67YxmqewACZAXdg vfg1ND0chkk7tFc5q3iPYrM= =zFFk -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] How to restrict simultaneous phone registrations
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jonathan k. Creasy wrote:
I apologize if this information is posted elsewhere. Unfortunately I
haven't found it yet if it is. I'm not familiar with the channel
counting features could you please explain? Also, how are you tagging
the phones to account codes?
You can limit calls using the set/check group commands.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
Account codes are set either by using the Set function or the
accountcode= property in the SIP/IAX conf files.
-Jonathan
Exactly, I'll post a sample dialplan. This dialplan is for ASTPP but
should give you the idea.
# exten = _1XX,1,Set(GROUP()=${ACCOUNTCODE})
# exten = _1XX,2,AGI(astpp-authorize.agi,${ACCOUNTCODE},${EXTEN})
# exten = _1XX,3,GotoIf($[${CALLSTATUS} = 0]?60) ; Checks
if account has sufficient funds
# exten = _1XX,4,GotoIf($[${CALLSTATUS} = 1]?70) ; Checks
if the phone number exists
# exten = _1XX,5,GotoIf($[${CALLSTATUS} = 2]?80) ; Check
if account exists
# exten = _1XX,6,GotoIf($[${GROUP_COUNT()}
${MAXCHANNELS}]?90) ; Verify number of outgoing channels
#
; assigned to account.
# exten = _1XX,7,Set(GROUP(${TRUNK1}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,8,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK1_MAXCHANNELS}]?10)
# exten = _1XX,9,Dial(${LCRSTRING1}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,110,Busy
# exten = _1XX,10,Set(GROUP(${TRUNK2}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,11,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK2_MAXCHANNELS}]?13)
# exten = _1XX,12,Dial(${LCRSTRING2}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,113,Busy
# exten = _1XX,13,Set(GROUP(${TRUNK2}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,14,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK3_MAXCHANNELS}]?16)
# exten = _1XX,15,Dial(${LCRSTRING3}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,116,Busy
# exten = _1XX,16,Set(GROUP(${TRUNK4}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,17,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK4_MAXCHANNELS}]?19)
# exten = _1XX,18,Dial(${LCRSTRING4}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,119,Busy
# exten = _1XX,19,Set(GROUP(${TRUNK5}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,20,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])-OUTBOUND}
${TRUNK5_MAXCHANNELS}]?22)
# exten = _1XX,21,Dial(${LCRSTRING5}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,122,Busy
# exten = _1XX,22,Goto(100)
# exten = _1XX,60,Congestion ; '0' Tells them they do not have
enough money
# exten = _1XX,61,Hangup
# exten = _1XX,70,Congestion '1' Bad Phone Number
# exten = _1XX,71,Hangup
# exten = _1XX,80,Congestion
# exten = _1XX,81,Hangup
# exten = _1XX,90,Congestion; Their outgoing channel limit
is full already
# exten = _1XX,91,Hangup
# exten = _1XX,100,Congestion; No Route Available
# exten = _1XX,101,Hangup
Some of the group counts are for outgoing trunks. It's just the first
one that you need.
- --
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] How to restrict simultaneous phone registrations
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Here's how I do it. I have the phones tagged to accountcodes and I use the channel counting features of asterisk to limit an accountcode to X number of simultaneous calls. Darren Wiebe [EMAIL PROTECTED] Bryan Mahin wrote: Lol.. To an extent I agree. But I feel the best way is to find a way to block the problem completely. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, April 05, 2006 10:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How to restrict simultaneous phone registrations I say just bill the user at extension 333 it's his responsibility to keep the login info private. If he disputes it, refund the first time then change the password to something really complicated then start billing him if it keeps happening after that! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan Mahin Sent: Wednesday, April 05, 2006 10:50 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] How to restrict simultaneous phone registrations :) I should rephrase my question. And included a bit more information on what I am trying to accomplish. Solution 1 (preferred) I am working on an asterisk installation where most end users will use softphones. If I am not able to lock down calling to one phone at a time, the end users will share their login information with friends, family, neighbors, and the some girl they meet on myspace. Currently, I am able to register two phones at separate locations with the same account on each phone and make concurrent calls. For example, If I login extension 333 at location A, and 333 at location B, simultaneous calls can be placed from both phones at the exact same time. I only want calls placed from extension 333 to work from either A or B not A and B concurrently. Here is my ideal solution. Location A wants to make a call, but location B has a call in progress. Location B has to either close their phone, or hang up before Location A can make the call. OR.. Solution 2. :) A way I can distinguish in my CDR the IP address or some other recognizable difference between the two locations when they make concurrent calls using the same extension. The complication here is; I can currently the log IP addresses, but as the end phones are on the internet, Nat'd, and I am using a siparator for traversal. As a result, my logs show the IP address of the siparator and I don't have any other data to distinguish the end phones. OR.. Solution 2.5 One thought I've had is to send logs from the siparator to a syslog server, parse them, hunt for simultaneous calls placed by the same accounts from different locations, and bill the end users accordingly. But I really dislike this idea as no one likes to be hit with surcharges. Any help or insight is greatly appreciated. Thanks again, Bryan Mahin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, April 05, 2006 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to restrict simultaneous phone registrations Bryan Mahin wrote: Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? This is a non-issue, because a second registration to the same account will override and previous registrations for that account. ___ - -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFENJMw4DADnh+tnOQRAnA6AJ9WPEQKXAVidz7g6aXkIbeCqD2LfwCdF7yd f3ImomYaAAikmfoocM76Pdo= =9FdV -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] morcdr v0.1 released
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
It looks nice. I have a suggestion and I hope the Asterisk-stat
author won't mind. Have you ever considered contributing the code to
his project? I've dealt with him and he seems to be very reasonable.
IMO it's really nice to avoid the duplication of effort. Unless, of
course, you did all the coding for this yourselves. If there is a
major disagreement or problem with it I can see branching other than
that it seems unfortunate.
Just my $0.02CDN.
Darren Wiebe
[EMAIL PROTECTED]
Mindaugas Kezys wrote:
!-- /* Style Definitions */ p.MsoNormal, li.MsoNormal,
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color:windowtext;} @page Section1 {size:595.3pt 841.9pt;
margin:3.0cm 1.0cm 2.0cm 3.0cm;} div.Section1 {page:Section1;} --
CDR Stats Analyzer and Report generator
It's a rework of famous Asterisk Stats written by Areski.
The main goal for this project is to concentrate more on PDF
reports (managers love them!).
Later more functions will be added. Please test it and send
suggestions how to improve it.
Licence: GPL
Examples, demo and more info on homepage:
http://www.paskambink.lt/mcc
Regards,
Mindaugas Kezys
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Re: [Asterisk-Users] Callback auto dialing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have a script that I use for conference callbacks. Here, I go. In an email 5 minutes ago I criticize branching products and I'm giving a link to a hack of somebody elses work. :-( Anyway, here's the link to what I use. http://www.astpp.org/index.php?n=ASTERISK.Code Eric Wieling, if you want the changes I made you're more than welcome to them and the sound files also. :-) Darren Wiebe [EMAIL PROTECTED] Cosmin Prund wrote: Hello everyone. This is an other question from a relatively newbie. I'd like to provide auto callback ability for my *. From my mobile I want to be able to call a number on the * and have it call me back on my mobile. I know how to generate a .call file from a script and I know how to call a script from the dialplan (in order to get the .call file generated). I also found the scripts on www.voip-info.org on callback voicemail but what I want is not voicemail. I just want to talk to the * and use it's much lower rates! What I do not know is what to write in that call file so I'll get an IVR when I answer the phone, not Voicemail or an other channel. It seems that call files are designed to connect one channel to an other channel or one channel to an application. But I don't want to connect to an application like Voicemail, I want the system to behave as if I called the other way around and ended up into an arbitrary context. Thanks for any help, Cosmin Prund, Romania ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEMbZX4DADnh+tnOQRAoAZAJ95eBmdNrfQWRRjcUfimNsvykQiqgCePn8a Kb9MuDiZZQqJmGEcTv46Alw= =iXQU -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] DID billing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, you can do that using ASTPP (www.astpp.org). You can bill DIDs per month and map them to the appropriate ATA device using the gui. Darren Wiebe [EMAIL PROTECTED] Bernard Cresencia - CrossNet International wrote: Hi all, Aside from the obvious answer write your own, anyone know of any DID billing software out there? I'm looking to possibly resell som DIDs to people and want to charge some per/minute rate when forwarding to their ATA devices. I can pull the data off mysql and create invoices but I'm kinda looking for an app similar to ASTCC, A2Billing, etc. where everything happens real-time. I looked at A2Billing but the software doesn't support billing if calling an IAX friend. I also looked at MCC billing but I can't seem to figure it out. Any help 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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEMf1t4DADnh+tnOQRAtu4AJ4sujwdCbB/CvkqarVJTLm2sdQXSACfbP0s WeC1xGAUFHr1KE/UPIKXge8= =xlh4 -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] Asterisk billing from CDR database
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Disclaimer: astpp is my software. :-) It's quite easy to do this with astpp. Depending on exactly what you want, there are a few ways to do it. Drop a note on the astpp forum (www.astpp.org/forum) or on the astpp mailling list if you're interested. Darren Wiebe [EMAIL PROTECTED] Chris Mason (Lists) wrote: I am copying the Master.csv file to another server and importing to mysql. I am looking for a simple billing application that will produce a bill for a give account code for a give period, based on a rate table. Is this available? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEJXC64DADnh+tnOQRAsiJAJ4wdn0PDP8ac0tyt92kxYwI2eE2qgCeIP5e /FOvPyAut5Psfk54Xqc1bag= =JftS -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] Asterisk billing from CDR database
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I didn't want to bore everybody with the details but I'll try. :-) A few questions/comments: 1. Are you wanting invoices to print or email or do you want something that keeps track of a prepaid balance? We can do both. ASTPP itself will handle the second and I've been using OSCommerce for invoice generation and presentation. If needed, we can easily build invoice generation into ASTPP I just haven't gotten that far yet. I have a user who has done it who could probably be prevailed upon to share at least some of the code. 2. Account payments... ASTPP does not have the capability at present to process credit cards or Paypal. This is another which I have been using oscommerce for. You can sign up for a voip account and refill your account using our oscommerce plugin. If I missed something let me know. Darren Wiebe [EMAIL PROTECTED] Chris Mason (Lists) wrote: Darren Wiebe wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Disclaimer: astpp is my software. :-) It's quite easy to do this with astpp. Depending on exactly what you want, there are a few ways to do it. Drop a note on the astpp forum (www.astpp.org/forum) or on the astpp mailling list if you're interested. I don't suppose you could just tell me? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEJYso4DADnh+tnOQRAoEvAJ4/rFHbqHIr6XCY6QG8+IeAAMGObgCeLFVX mMfgWkJpwF54w8VEapnPhcc= =orxL -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] asterisk billing
-BEGIN PGP SIGNED MESSAGE-
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We will be able to do that using ASTPP whenever I have time to spend
on Local channel problems that have cropped up for me. That is part
of the reason that I have support builtin for the local channel. I
guess it doesn't actually drop them back but it does keep the call on
the local host.
Darren Wiebe
[EMAIL PROTECTED]
Jeremy wrote:
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Does anyone know any asterisk billing utilities that would drop the
caller back to your own IVR after authentication and still log
time used. . . no dial out needed.
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Re: [Asterisk-Users] Grabbing the billsec and duration after a hangup.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have a perl app that listens for hangups and then grabs the call out of the database using the uniqueid. Maybe not the neatest way but it works well. Darren Wiebe [EMAIL PROTECTED] Mark Ackroyd wrote: The reason for it being 0 is because as long as you sit on the h extension the call is not yet done, therefore asterisk has no clue what those valuse are. If you use the h extension then you are messing up the CDR. So how can I tell it the call is complete and give the CDR values? Is it just not possible? Mark ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEH1hU4DADnh+tnOQRAh9NAJ4t+T0dteQQouh8LrsrHy6b27GEgACfcK07 Nc/6CfM4twpq6nb+TVk+Rnc= =RxIJ -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
[Asterisk-Users] Local Channel
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm using the Local channel in an app of mine and I'm finding that the app is being cut out of the call path. You used to be able to avoid this using the \n command but that doesn't seem to work any more. This is on a recent version of Asterisk. Any comments/suggestion? Darren Wiebe [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEHjLG4DADnh+tnOQRArn+AJ0dx9fncjX77QVtP0VzCXqa2i0BXwCdFv1v 0UQ9s6cloDFZJwIiBWJe/Hg= =fi8U -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] Re: Asterisk Question
my ( $var1, $var2, $var3 ) =
@ARGV;
and so on and so forth.
Good Luck
Darren Wiebe
[EMAIL PROTECTED]
Paul Hales wrote:
Thanks for this example - it has really got me started!
Short question - how can I put a variable into my perl script?
I imagine it's something like
exten = 780,1,AGI(agi_ret_val2.pl|${back})
But how can I get my perl script to pick this value up?
Again - thanks to everyone who has helped me with this.
later,
PaulH
On Tue, 2006-02-28 at 11:25 -0800, Michael Collins wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk Question
I was going to see if I can execute a bash script as an AGI - just
looking
around the internet for examples at the moment.
Anybody got an example spare?
I'm just a bit stuck on how to start this, but I am quite comfortable
writing asterisk dialplan stuff and bash scripts
later,
PaulH
Paul,
I'm a Perl guy myself. Here's a simple dialplan extension and AGI
script written in Perl and using the very cool Asterisk::AGI module:
; AGI test
exten = 555,1,Noop(Starting AGI test)
exten = 555,n,Answer
exten = 555,n,Wait(1)
exten = 555,n,Playback(beep)
exten = 555,n,AGI(agi_var_test.pl)
exten = 555,n,SayDigits(${EXTERN_VAR})
exten = 555,n,Wait(1)
exten = 555,n,Playback(beep)
exten = 555,n,Hangup
Here's the Perl script:
#!/usr/bin/perl
#
# agi_var_test.pl
#
# Reads in info from file /etc/group
# assigns asterisk GID to Asterisk variable EXTERN_VAR
#
use strict;
use warnings;
use Asterisk::AGI;
# the AGI object
my $agi = new Asterisk::AGI;
# pull AGI variables into %input
my %input = $agi-ReadParse();
my $infile = '/etc/group';
open(FILEIN,,$infile) or die $infile - $!\n;
while(FILEIN) {
chomp;
next unless m/^asterisk/;
my @REC = split :,$_;
print STDERR agi_var_test.pl: Setting EXTERN_VAR to $REC[2]\n;
$agi-set_variable(EXTERN_VAR, $REC[2]);
last;
} # while(FILEIN)
close(FILEIN);
Basically the script just parses /etc/group until it finds the asterisk
entry. It then parses the data line and extracts the GID. Finally, it
prints the value to STDERR (for debugging purposes) and then assigns the
value to EXTERN_VAR.
This is more a proof-of-concept than anything else, but it does show the
value of AGI and Asterisk::AGI.
-MC
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] AGI Scripts Terminate too Soon
In that case, asterisk sends -HUP to the agi script (I believe). Darren Michael Collins wrote: If that's true, why does dial() return control to the script when the callee hangs up? Doug, if I understand the AGI limitation correctly, the 'dead' in DeadAGI() refers to the other end of a dial() connection. I *think*, but I'm not positive on that. Does anyone know the answer to this one? Thanks, MC ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] billing - different tarif per phone
I think that the feature you're looking for is called pricelists in ASTPP but I could misunderstand what you want. Feel free to post the question either on the astpp-users mailing list or the astpp forum. Visit www.astpp.org for more info. Darren Wiebe [EMAIL PROTECTED] Pavel Jezek wrote: Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 phone number pattern match query
What are the contents of your database? If you can put in a regex expression then I can tell you exactly how to do it, otherwise I can tell you close. In ASTPP, I'm doing it similar to how ASTCC does it. I'll lay it out here: Pattern field in CDR ^1403.* will match anything beginning with 1403. Let's say you had dialed 1403888. You would have a mysql query like this: SELECT * FROM list WHERE '1403888' RLIKE pattern ORDER by LENGTH(pattern) DESC /blatant plug starts/ Are you building a billing system? If so, have a look at www.astpp.org, it has all this sort of stuff in place already. /blatant plug ends/ Hope this helps Darren Wiebe [EMAIL PROTECTED] Damon Estep wrote: Does anyone have a mysql query that will compare a number from the asterisk cdr to a table of international country+city codes to determine the closest match? The two fields are; 1. Asterisk mysql cdr ‘dst’ field – sample record value ‘011441316551212’ 2. rate table data like this DialPattern 011447977 011447979 011447980 011447981 011447984 011447985 011447986 011447987 011447988 011447989 011447990 011448 011449 01144 The goal is to find the _/longest/_ matching record from the rate table for each dialed number. In this case ‘01144’ I am not a mySQL expert (obviously), my limited SQL experience is with MS SQL where stored procedures and views are an option. This is with mySQL 4.x, so no views. Something like this Select dialpattern from rates where left 5 match left 5 of dst Order by length of dialpattern, descending Compare dialpattern to the first x number of digits from dst where x = the length of dial pattern The first match (when ordered by length descending) is the correct result (longest match) Too bad mySQL does not understand English J ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Billing inbound calls per minute
Hours of struggling later, I have found the problem. Here is the
correct format for those outgoing calls.
SIP/[EMAIL PROTECTED]||L(54081429:6:3)|Hj
I'll try to get a patch done up one of these days.
Darren Wiebe
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
I've been playing with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
Michiel van Baak wrote:
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
On Monday 06 February 2006 09:25, JP Carballo wrote:
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
On Saturday 11 February 2006 06:32, Darren Wiebe wrote:
Are you running a relatively recent version of ASTCC? Say within the
last 6 months. The answeredtime = 0 bug was supposed to have been fixed
by http://bugs.digium.com/view.php?id=4300 Unless something has changed
in Asterisk that affects this
Thanks Daren,
Yes, my version of astcc is the most recent one.
Asterisk-1.2.4
I have found you patch 0004300 from 16 May 2005.
Probably it's time to reverse it back since something has changed
in Asterisk that affects this... as you said.
My observation is:
If I keep:
$dialstr = Local/[EMAIL PROTECTED]{path}|30|HL/n( . ($maxtime * 60 * 1000) .
:6:3);
Either the billseconds is empty(when dial out through Local), either there is
aZOMBIE when dialing in.
I put back the dialstring to:
Local/$phone/$res-{path}|30|HL/n( . ($maxtime * 60 * 1000) .
:6:3);
The only difference that it looks only for is a default context.
extensions.conf
[inbound]
; 10 digits DID = _XX = cardnumber
;
exten = _XX ,1,Answer()
exten = _XX ,n,Set(DB(RCID/${CALLERIDNUM})=${CALLERIDNUM})
exten = _XX ,n,Set(realcid=${DB(RCID/${CALLERIDNUM})})
exten = _XX ,n,Noop(${REALCID})
;exten = _XX ,n,Set(TIMEOUT(digit)=4)
exten = _XX ,n,Set(CALLERID(number)=${EXTEN})
exten = _XX ,n,Set(CALLERID(name)= ${REALCID})
;exten = t,3,Goto(h|1)
;exten = _XX 2,Goto(s|1)
;exten = s,1,Wait,1 ; is this preventing HUP?
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${CALLERIDNUM},4)
; must be h,1 as per Michiel van Baak note(above).
exten = h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten = 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include = internal
[personal]
exten = t,1,Hangup
include = inbound
Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but looong problem - Timeout about 10
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.
Sorry for the long explanation.
What do you think? Is there something suspicious in
that solution?
Thanks,
benchev
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[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
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Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls
Well, I'm not real sure on whether I like the idea or not bug Anyway, here is an app that I wrote for something similar to this. It was for notifying customers of events,etc. http://www.astpp.org/index.php?n=Misc.AutoDialOut Darren Wiebe [EMAIL PROTECTED] Ron Senykoff wrote: Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on nerdvittles: http://mundy.org/blog/index.php But I'm not sure that this actually does concurrent calls. I'm thinking my best bet is writing some fast agi to parse a mysql database, then create call files. Use asterisk manager interface to monitor calls and that way I can keep the preset concurrent limit. Any ideas? TIA! -Ron ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] IP Authorization
It's part of ASTPP. It is in astpp -head ready for testing. Darren Wiebe [EMAIL PROTECTED] Sam Tam wrote: When will it be ready ? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Saturday, February 11, 2006 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Authorization I'm doing it similar to what the posted showed today. Then I'm calling an agi script (Maybe not the nicest way) that checks to see if the IP is allowed and sets the accountcode for the call. Darren Sam Tam wrote: Can you be more detail about the setup? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: Friday, February 10, 2006 4:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Authorization Sam Tam wrote: I think this is a question that has been discussed before. But you see nowadays most carriers will provide thing like SIP using IP authorization rather than username and password and I am now wondering whether Asterisk can do something like that or not? In the voip channels as well as in manager you can set ACLs for the connections you define. /O ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] IP Authorization
I'm doing it similar to what the posted showed today. Then I'm calling an agi script (Maybe not the nicest way) that checks to see if the IP is allowed and sets the accountcode for the call. Darren Sam Tam wrote: Can you be more detail about the setup? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: Friday, February 10, 2006 4:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Authorization Sam Tam wrote: I think this is a question that has been discussed before. But you see nowadays most carriers will provide thing like SIP using IP authorization rather than username and password and I am now wondering whether Asterisk can do something like that or not? In the voip channels as well as in manager you can set ACLs for the connections you define. /O ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Billing inbound calls per minute
Are you running a relatively recent version of ASTCC? Say within the
last 6 months. The answeredtime = 0 bug was supposed to have been fixed
by http://bugs.digium.com/view.php?id=4300 Unless something has changed
in Asterisk that affects this
[EMAIL PROTECTED] wrote:
On Monday 06 February 2006 09:25, JP Carballo wrote:
Michiel van Baak wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
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[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] IP Authorization
This is hopefully on topic. I'd like thoughts on this. I'm looking at doing some dialplan work which would grab the sip devices IP number. If that ip number is in an allowed list, the call would be allowed to go through otherwise congestion would be passed. Any thoughts? Darren Wiebe [EMAIL PROTECTED] Olle E Johansson wrote: Sam Tam wrote: I think this is a question that has been discussed before. But you see nowadays most carriers will provide thing like SIP using IP authorization rather than username and password and I am now wondering whether Asterisk can do something like that or not? In the voip channels as well as in manager you can set ACLs for the connections you define. /O ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] ASTPP
This doesn't really belong on the asterisk-users list. ASTPP has it's own mailing list. This can be found @ www.astpp.org. I, or someone else will be happy to help you either there or on the forums. On your 1st post please mention what version of ASTPP you are using. Thanks, Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.astpp.org Ronald Ramos wrote: Hi Sir, My problem is when I click on pricelist, i have an error there's something wrong on the pricelist database. When I looked at the database and search for a table called pricelist there's nothing there. I foolowed the querires on the the structure but also found any query that creates the pricelist table. Is the pricelist going to be created at the start or after I've setup everything? Thank You Regards, Ronald JP Carballo wrote: Under Rates click on - Pricelists then Add... ___ --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 Prepaid Solution
JP Carballo wrote: Ronald Ramos wrote: Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald Hi Ronald, Check the prepaid applications here for ideas: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications ASTPP, which is based on ASTCC is highly recommended. http://www.aleph-com.net/astpp Myself, I've implemented what you aim to do using ASTCC hooked to the shopping cart Virtuemart/Joomla. Customers register through Virtuemart/Joomla, then a card is created on ASTPP. When they buy a refill card through the store, their account is credited. That's cool! I have been working on integrating ASTPP with oscommerce. I had hoped to have a release out for Jan 1st but I am behind. Check out the astpp demo. www.astpp.org As for * on ser, you may want to visit : http://www.voip-info.org/wiki-SIP+Express+Router Good Luck, -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 Prepaid Solution
I know, the recurring charge stuff is a pain right now. I have added support into ASTPP to do recurring charges. The problem is actually getting the money. I'm going to try to get something designed for that also but for now it can apply a recurring charge to a customers account. Maybe I'll make progress on the weekend. :-) Darren Wiebe [EMAIL PROTECTED] JP Carballo wrote: Darren Wiebe wrote: JP Carballo wrote: Ronald Ramos wrote: Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald Hi Ronald, Check the prepaid applications here for ideas: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications ASTPP, which is based on ASTCC is highly recommended. http://www.aleph-com.net/astpp Myself, I've implemented what you aim to do using ASTCC hooked to the shopping cart Virtuemart/Joomla. Customers register through Virtuemart/Joomla, then a card is created on ASTPP. When they buy a refill card through the store, their account is credited. That's cool! I have been working on integrating ASTPP with oscommerce. I had hoped to have a release out for Jan 1st but I am behind. Check out the astpp demo. www.astpp.org Thanks Darren! The backend is really a hybrid of ASTCC and ASTPP's calling card part. I remember telling you about a month ago that I had osCommerce setup and was trying to get either ASTCC or ASTPP to work with it. I gave up, lol. I was spending far too much time patching osCommerce. Imho, Virtuemart/Joomla is much easier to customize and maintain. The only hurdle I see is Virtuemart's current inability to handle recurring charges or monthly payments. Once a module exists for that, I can use ASTPP's postpaid capabilities. -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] need help
Have you settled on a calling card application yet? There are a host of different options. I, of course, recommend astpp. :-) The wiki will have much of the info you will need. Darren Wiebe [EMAIL PROTECTED] Dirgan Putra wrote: Iam new in asterisk user, can helpme to install asterisk for applications callingcard ? current ialready install asterisk with mysql db and already connected, and next i dont know how to create as calling card applications, adn how other how to setup using SIP sofphone if iam using expresstalk softphone, if i want use as callingcard applications. pls help thanks Dirgan Meet your soulmate! *Yahoo! Asia presents Meetic* http://us.rd.yahoo.com/prm/pers/mt/yma_sg/tgl/*http://asia.yahoo.com/meetic - where millions of singles gather ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] New Freelance Site for Asterisk Consultants and Those who Need Projects Done
Cool! I just tool a look at it looks like you did a great job!! Darren Wiebe [EMAIL PROTECTED] Steve Totaro wrote: Sorry if this is slightly off topic but it does pertain to Asterisk Users as well as the biz list. Also, sorry if it is a double post but the first one never made it to the list for some reason. Hello all, I have created a beta site for Asterisk Gurus or Consultants to bid on projects posted by customers needing to have work done. It is very similar to scriptlance or any of those other sites but it is dedicated to Asterisk and related issues so hopefully only really qualified Asterisk consultants will bid on your projects. If you post at one of those other sites, you wind up with 99% of the people who bid unable to complete the project and they waste your valuable time. Asterisk is a very specialized skill and with our rating system, we can quickly identify who the good Asterisk Gurus are and not waste time with the wannabes. This also seems to be a very good replacement for the Bounty system on www.voip-info.org http://www.voip-info.org/ . I am sure we can figure out how to split costs owed to the Asterisk Guru between customers. It is VERY beta right now but I think it is also fully functional. Any reference to payments, deposits, $$$, etc can be ignored. The service is free for now and will stay that way for at least the next six months. However, I am may add a PayPal donation link since this is certainly not free for me. Of course there is no obligation to donate but I would appreciate it. Heck, if there are enough donations then the site could remain free permanently. There are some small issues since the script is wrapped in another script, but I am aware of this and will find a fix shortly. Besides that, I could use any input on usability, additions, categories not listed, or whatever jumps to mind. I will also be adding a section to post resumes and other permanent job postings. Please test it out and let me know what you think. http://www.asteriskhelpdesk.com Thanks, Steve Totaro ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Dialer
http://www.astpp.org/index.php?n=Misc.AutoDialOut I put together what I have on that site. Darren wiebe [EMAIL PROTECTED] Steve Totaro wrote: Darren, I am interested in your project. Let me know if I can help you test. Thanks, Steve -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, January 06, 2006 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialer If this or any other example is available, I would be most thankful to have it. I got the go ahead on this project to day so now I have to start seeing how to do this. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Tuesday, January 03, 2006 5:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dialer I'm supposed to have a mostly canned script that will do this done already. It will pull the list of people to call out of a db and play them the file specified in the db table. Contact me offlist if you're interested. It will be done real soon but I'm not done testing yet. Darren Wiebe [EMAIL PROTECTED] Kerry Garrison wrote: You actually aren't far from it. If the system only needs to play the same file to each person, a simple script can be used to pull from a database and create call files. Asterisk will use the call files to place the calls and play a sound. A few minutes of searching on that should get you started. I haven't seen anyone else have a canned script ready to go, but would like to know if anyone does. -Kerry *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Wiley Siler *Sent:* Tuesday, January 03, 2006 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Dialer Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley --- - ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Some advice on routing DID's
I have written an agi script that I use for that. Then I can just have a list of dids and extensions in a db. Tom Vile wrote: Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Some advice on routing DID's
Just grab the script. I can help you with it off the mailing list if you like Darren Tom Vile wrote: Do I need to install the complete ASTPP package or just utilize your AGI script with the context for AMP? Thanks On 1/7/06, Tom Vile [EMAIL PROTECTED] wrote: It's funny you mentioned that Darren, I was looking at your scripts today. I will evaluate it some more. On 1/7/06, Darren Wiebe [EMAIL PROTECTED] wrote: I have written an agi script that I use for that. Then I can just have a list of dids and extensions in a db. Tom Vile wrote: Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Using local\number
Local/[EMAIL PROTECTED] Try putting the context on. I do this all the time in callfiles. Darren Wiebe [EMAIL PROTECTED] Matt wrote: Hi, What do I have to do to get local\number to work in a context? It works from my [from-internal]... however from subcontexts it does not work: Jan 6 15:55:32 VERBOSE[20237] logger.c: -- AGI Script Executing Application: (Dial) Options: (Local/570323) Jan 6 15:55:32 NOTICE[20237] chan_local.c: No such extension/context [EMAIL PROTECTED] creating local channel Jan 6 15:55:32 NOTICE[20237] app_dial.c: Unable to create channel of type 'Local' (cause 0 - Unknown) Jan 6 15:55:32 VERBOSE[20237] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Jan 6 15:55:32 DEBUG[20237] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. I'm dialing this as: Local/570323 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Dialer
I'll try to finish this up tonight and post back once I'm done. Darren Wiebe [EMAIL PROTECTED] Wiley Siler wrote: If this or any other example is available, I would be most thankful to have it. I got the go ahead on this project to day so now I have to start seeing how to do this. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Tuesday, January 03, 2006 5:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dialer I'm supposed to have a mostly canned script that will do this done already. It will pull the list of people to call out of a db and play them the file specified in the db table. Contact me offlist if you're interested. It will be done real soon but I'm not done testing yet. Darren Wiebe [EMAIL PROTECTED] Kerry Garrison wrote: You actually aren't far from it. If the system only needs to play the same file to each person, a simple script can be used to pull from a database and create call files. Asterisk will use the call files to place the calls and play a sound. A few minutes of searching on that should get you started. I haven't seen anyone else have a canned script ready to go, but would like to know if anyone does. -Kerry *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Wiley Siler *Sent:* Tuesday, January 03, 2006 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Dialer Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley --- - ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Dialer
I'm supposed to have a mostly canned script that will do this done already. It will pull the list of people to call out of a db and play them the file specified in the db table. Contact me offlist if you're interested. It will be done real soon but I'm not done testing yet. Darren Wiebe [EMAIL PROTECTED] Kerry Garrison wrote: You actually aren't far from it. If the system only needs to play the same file to each person, a simple script can be used to pull from a database and create call files. Asterisk will use the call files to place the calls and play a sound. A few minutes of searching on that should get you started. I haven't seen anyone else have a canned script ready to go, but would like to know if anyone does. -Kerry *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Wiley Siler *Sent:* Tuesday, January 03, 2006 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Dialer Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Is there a GUI for asterisk realtime
There is a web interface. It's pretty basic but you can find a demo here: http://dc.maxnet.ru/cpdemo/ I know the guy that owns it. Contact me if you're interested. It's payware. Darren Wiebe [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, Is there a GUI to manage the users in database (realtime) ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] CVS problem?
That is because they switched over to svn I belive. Darren Wiebe Colin Anderson wrote: cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel' Anyone else seen this? ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] CVS problem?
You know, that's right. I thought so too. I've been entirely unsuccessful getting cvs downloads but that could just be my luck. Merry Christmas Everyone, Darren Wiebe [EMAIL PROTECTED] Colin Anderson wrote: I thought they were going to run CVS concurrently for a while?? -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CVS problem? That is because they switched over to svn I belive. Darren Wiebe Colin Anderson wrote: cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel' Anyone else seen this? ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] SIP Subscriptions
http://www.voip-info.org/wiki/view/Asterisk+bounty Darren Wiebe [EMAIL PROTECTED] Douglas Garstang wrote: Digium needs people like me, if they read this list that is. They sure don't seem to be able to make real-world functionality decisions on their own. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 3:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions On 20/12/05, Douglas Garstang [EMAIL PROTECTED] wrote: So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned. Off you go to another product then. Close the door on the way out. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] SIP Subscriptions
Everybody is entitled to their own opinion. I believe Kevin Fleming indicated that the Digium todo list was flexible if enough $$$ of funding were involved. Maybe that would interest you more. Darren Wiebe [EMAIL PROTECTED] Douglas Garstang wrote: I don't think the bounties are worth the costs asociated with contracts, legal fees, international boundaries etc. -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions http://www.voip-info.org/wiki/view/Asterisk+bounty Darren Wiebe [EMAIL PROTECTED] Douglas Garstang wrote: Digium needs people like me, if they read this list that is. They sure don't seem to be able to make real-world functionality decisions on their own. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 3:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions On 20/12/05, Douglas Garstang [EMAIL PROTECTED] wrote: So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned. Off you go to another product then. Close the door on the way out. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] SIP Subscriptions
Douglas, the asterisk community is largely made up of volunteers. I have worked with volunteers in other places. Here is one thing they have in common. A volunteer does not usually feel obligated to put up with non constructive criticism. I smiled as I read some of your posts. You may not have meant them bad but they did come across rather strongly. To put it in the words of somebody I used to work with, He was talking to me like he though I cared. :-) I'm sorry but, very few people here really care whether or not you use asterisk. That is one difference between a project like this and a commercial system that you paid $. Your dealer may want to make you happy because that will put money in their pocket. I, quite frankly, don't really care if Asteirsk works for you. If you are having issues and the mailing list does not answer them for you, you have a couple of options. 1. Hire a consultant or programmer to fix it for you. 2. Try to hire Digium to fix it for you. 3. Find another application that works for you. Darren Wiebe [EMAIL PROTECTED] Douglas Garstang wrote: I don't know what the rules are for this list, but it wouldn't be much of a stretch to assume that personal attacks are grounds for removal. While we're at it, why does my reluctance to deal with contracts, legal fees and international boundaries make me a 'fuc*ing retard'? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tue 12/20/2005 7:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Subscriptions On 12/20/05, Douglas Garstang [EMAIL PROTECTED] wrote: I don't think the bounties are worth the costs asociated with contracts, legal fees, international boundaries etc. Right, Avaya is. This guy is a Fuc*ing retard. Douglas, did you see a doc the last few days??? ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Latest Source
: 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 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] astcc issue
You should be able to edit prices from within the routes page. However, you can't set different prices on different brands more accurately than by using markup. That is one of the reasons that I've branched / mostly rewritten the product. ASTPP, www.astpp.org, does provide support for doing this with prices but the calling card stuff is only in cvs yet. Darren Wiebe [EMAIL PROTECTED] jonny hashem wrote: Hi list: I need to create a routes list to specific card number wih different prices than the initial routes list ,because markup donot achieve my purpose and markup use for changing prices for all routes,and i need to change prices for specific routes. So is there any possible way to do that? Regards; jonny __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Astcc help
I think you have to run the update_database function. That code was written over a year ago and has not been touched since that I'm aware of. I suspect the Friends support should be moved over to realtime but. Darren Wiebe [EMAIL PROTECTED] Insider KT wrote: Hi. I am having some problem with Astcc. It works, but I would like to have IAX and SIP_Friends to work also. Hope someone here can help. In the web admin interface: I have put YES in : Enable Iax/Sip Friends DB (YES/NO) Then I have pressed create database and all is fine. The database is made, but without Iax_friends or Sip_Friends in it. If I then press Users_configure or Iax_Friends it says NOT CONFIGURED. I think I am missing something important here, but I don't know what. I find nothing after 3 hours of searching google. Does anyone here know what I am doing wrong ? Fredrik ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] ASTCC/ASTCC anything wrong with that?
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
$dialstr =
IAX2/$res-{path}/$phone|30|HL(
. ( $maxtime * 60 * 1000 )
. :6:3);
2. EVery trunk is closed lake that:
iaxprovider/
otherprovider/00
yetanother/
Q: see anything very wrong with that?
Thanks,
benchev
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Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] ASTCC/ASTCC anything wrong with that?
If it works fine, I can't think of any collateral damage. All that
affects is the way the string is put together.
Darren
[EMAIL PROTECTED] wrote:
Thanks.
It works fine. I was just curious about
any collateral damages.
Thanks again,
benchev
On Monday 12 December 2005 16:42, Darren Wiebe wrote:
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
$dialstr =
IAX2/$res-{path}/$phone|30|HL(
. ( $maxtime * 60 * 1000 )
. :6:3);
2. EVery trunk is closed lake that:
iaxprovider/
otherprovider/00
yetanother/
Q: see anything very wrong with that?
Thanks,
benchev
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Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?
You have to do that from the dialplan. I have a script that looks up
the DID in a database and sets the accountcode. It does some other
stuff also but that could easily be cut out. It's part of ASTPP. Drop
me a line if you need a copy.
Darren Wiebe
Matt wrote:
Hrmm that works except that my accountcode is not the extension of the
customer/user, but is a distinct accountcode (ID).
Oooo... you are setting the accountcode when you GET the
call. I guess I could do that... before I go to do too much work, is
there a way to get asterisk to know the accountcode for the inbound
call?
On 12/6/05, Andy Kuo [EMAIL PROTECTED] wrote:
I use SetAccount(${EXTEN}) when the extension gets the call. The original
dialed extension will be recorded as AccountCode in CDR, before the call is
forwarded. The 1st field in CDR will be the extension your customer, the
2nd will be the caller (source), the 3rd will be the forwared number.
It works for me pretty well.
Andy
On 12/6/05, Matt [EMAIL PROTECTED] wrote:
I want to allow my users to be able to
Call Forward Unconditional
Call Forward Busy
Call Forward No Answer
And curently I am doing this via my ATA and phone settings, however
this has the problem that when a call is forwarded it goes out without
an accountcode (Even though the ATA is forwarding the call), and hence
I can't track the call!
Can someone suggest a way to either fix this so that accountcodes go
into the CDRs when the ATA/phone forwards the call, or to do the three
forwarding types directly on asterisk?
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Re: [Asterisk-Users] callback script
You can leave the stuff in callback.agi the way it is.
[enhanced-outgoing]
exten = _1XX,1,Dial(SIP/000.000.000.000/${EXTEN})
exten = _1XX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1XX,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1XX,4,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _011.,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _011.,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
[meetme]
exten = 9928,1,setaccount(customer)
exten = 9928,2,Playback(you-are-being-connected-broadcast)
exten = 9928,3,AbsoluteTimeout(12600)
exten = 9928,4,Conference(conference1/M/1)
exten = 9929,1,SetGroup(customer)
exten = 9929,2,setaccount(customer)
exten = 9929,3,Playback(you-are-being-connected-to-the-service)
exten = 9929,4,AbsoluteTimeout(12600)
exten = 9929,5,Conference(conference1/L/1)
That is what I have. You would want to replace the meetme stuff with
whatever you want the other end to connect to.
Darren
wassim darwish wrote:
Hi:
Once i have seen the post of Darren Wiebe of
suggestion of a callback configuration in
extensions.conf and it was like this:
[callback]
exten =
_.,1,AGI(callback.agi,LAKEVIEW,1234567890,9998,,meetme,enhanced-outgoing)
But i didnt know what to add in meetme and
enhanced-outgoing contexts.
if any body knows about this configuration ,just show
me what to put in the meetme and enhanced-outgoing
contexts and what to edit in this part of callback.agi
script:
$outgoingclid = ;
$channel = ;
$context = ;
$church = ;
Regards;
wassim
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Re: [Asterisk-Users] prepaid application
Steve Totaro wrote: Hi All I am using prepaid auth (callingcards), the idea is for a prepaid support line. It is up and running but I have a couple of questions with regards to modifications I would like to make. When a user calls and they go through the process of entering their card number. They are then asked for a destination. What I would like to be able to do is not have it ask for a destination and automatically dial a number? I have done this exact setup for a prepaid information service. I did everything by editing astcc.agi. I also have it setup to handle three different languages. What I did for this was to edit the agi exec dial and hard code the dial number right in. I also modified the script to not disconnect a caller when their credit runs out since that is pretty rude. I'd probably do it this way too. You could get away without hardcoding the number to dial in if you wanted to do a little extra work in the dialplan. Darren Email me offlist for help. Thanks, 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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] launching 2 scripts
I'm not exactly sure what you're trying to do. I don't know which call back script you are using, but you should be able to set which context and extension you want the call connected to. I do that using a callback script that I found on the internet somewhere. I did some work to it and it is available @ www.aleph-com.net/astpp. This is the way I run that one: [callback] exten = 1,1,AGI(callback.agi,ACCOUNTCODE,99,,9959,meetme,enhanced-outgoing) 99 can be a CID number. If that number dials in it gets connected to instead of 9959 meetme is the context to throw the call into when it's connected enhanced-outgoing is where I send the outgoing calls through. I use the local channel. Make sense? Darren Wiebe [EMAIL PROTECTED] chawki hammoud wrote: Hi: i tried to lauch the callback.agi script and astcc.agi script together but i failed to do that ,i tried this at extensions.conf: [incoming] exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,DeadAGI(callback.agi) exten = s,4,DeadAGI(astcc.agi) exten = s,5,Hangup i tried to make astcc.agi launch when the call answered when it callback but i failed. __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Really lightweight itemised billing
Do you have accountcodes in the database? If you do, you could use astpp quite easily. We could cut out most of the functionality for you. Right now I don't have a way to search by date but that would be failry easy to add and I will be working on it soon anyways. Drop me a line if you want or visit www.aleph-com.net/astpp Darren Wiebe [EMAIL PROTECTED] Chris Bagnall wrote: Good morning all, I'm trying to find an application that'll do really lightweight billing for Asterisk CDRs. On our asterisk servers deployed at people's offices, we have CDRs being logged to PostgreSQL, which can then be analysed by the staff at those offices using a PHP-based CDR analyser. This works fine for legitimate use verification (it's easy to spot people making hours of phone calls to their girlfriend's mobile, for example), but it doesn't provide billing verification. All I'm looking to do is parse the CDRs for a given date range, lookup each dialled number in a table to get its rate, then present a formatted list (or even a .csv) of the person dialling (accountcode), time/date of call, duration and total cost of call. All of the billing applications I've seen so far are either 1) really heavyweight designed for calling card or other charging purposes, or 2) want me to modify the asterisk configuration to use their AGIs for dialling. It's an overkill for what I'm after. Before I go and write some PHP scripts to do what I'm after, has anyone already done this and have some scripts they want to share? :-) Thanks in advance. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC - card in use
Ronald Wiplinger wrote:
Is there a solution for the problem that the card in use flag is set,
after the user hang up?
Yes, there is a patch. This was fixed in cvs quite a while ago.
Put this:
$SIG{HUP} = 'ignore_hup';
sub ignore_hup {
print STDERR \nHUP received!\n\n;
}
just after the use POSIX qw(ceil floor); line
The flag remains set, if the user hang up, after the price for the
call will be announced.
It is bad (for the business), because this happens most of the time
only for NEW users!
Solutions?
1. Do we need the flag at all, if we use the phone number as card
number anyway? If we can use the phone number as card number, we could
omit it. However, I plan that I will have other premium features,
where I need to punch in a card number!!!
The flag keeps one card from having more than 1 simultaneous call. If
you don't want to use it, a bit of work in astcc.agi would disable it.
2. Could we use a extension number to reset the flag?
You could this this using an agi script.
3. Could we use a cron job to reset the flag, if the extension number
is not in a call? This one sounds for me most reliable, but at the
moment the most complicated one to figure it out by the cron job and
reset it from there.
This would be fairly easy. You would need a script that ran the
appropriate sql command when called.
Is a solution available?
Yes, it is.
Good Luck Ronald, I haven't talked to you for a long time. :-)
Darren Wiebe
[EMAIL PROTECTED]
bye
Ronald
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Re: [Asterisk-Users] AGI and AUTOHANGUP
This is easy. :-) How are you creating the call? From an AGI script? Here's how I do it. $maxtime = $timelimit * 1000; #$maxtime is maxlength in seconds $timelimit = |30|HL($maxtime:6:3); #This will provide a warning @ 60 and 30 seconds. return $timelimit; Append $timelimit onto the end of your dialcommand. You can look at the code for ASTCC in the asterisk cvs or look at astpp-callingcard.agi in the cvs code available @ www.aleph-com.net/astpp Darren Wiebe [EMAIL PROTECTED] Innocent Evil wrote: Comeo'n AGI guys.. Please say something. Hi, Using AUTOHANGUP, I can force a call duration within a time limit. I would like to playback a message before 1 minute of autohangup. How can I accomplish it? Would anybody please give me right direction. Thanks, You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[2757]: Failed to write frame
I just upgraded to 1.2 and that fixed the problem for me. Darren Wiebe [EMAIL PROTECTED] Abdock wrote: Hello, Getting this error and the audio is too low, file.c:550 ast_readaudio_callback: Failed to write frame How to get correct this ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] set local area code
Use a line like this in your dialplan. I'll post a sample out of mine.
exten = _NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1780${EXTEN}
That line is setup so any 7 digit numbers will be marked as belonging to
1-780.
Darren Wiebe
[EMAIL PROTECTED]
Jason Brashear wrote:
How do you set it up so that you don't have to dial you area code ie 512 ?
-J
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Re: [Asterisk-Users] CallBack Suggestion
My use of it is slightly strange but I'll post it here. exten = _.,1,AGI(callback.agi,LAKEVIEW,1234567890,9998,,meetme,enhanced-outgoing) LAKEVIEW - The accountcode to bill the call to 124567890 - Will get mapped to [EMAIL PROTECTED] All other numbers will get mapped to [EMAIL PROTECTED] All outgoing calls will be placed through the Local channel in context enhanced-outgoing. Hope this helps Darren Wiebe [EMAIL PROTECTED] Musaluke AK wrote: Darren, An example how to call that callback.agi script? The script iself does not have usage info. Thanks Anthony Darren Wiebe wrote: Hello. You should not need any special hardware for callback. You will (obviously) need card to connect your box to the pstn. Do you have something setup with freeradius already? If not, you could quite easily setup something like this with ASTCC. I have a callback script @ www.aleph-com.net/astpp. Somewhere there. It is way more complicated than you need but you can cut out all the user interaction stuff. Darren Wiebe [EMAIL PROTECTED] Abdul Lateef wrote: Hi friends, I am new in asterisk, i came for CallBack purpose, i read from Voip-info.org aboue callback with asterisk and i am near to collect all information about to start developing callback system. Just i have a samall question, Is Callback needs some special hardware? i have my PSTN phone number i want to call this number after two ring the call will be disconnect and the Callback will start to call back to the caller ID and it should prompt to enter pin id which will authunticate via freeradius.if the authuntication is valid it will give some beep for dialing the international number. Any kind of suggestion will be hearty appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallBack Suggestion
Hello. You should not need any special hardware for callback. You will (obviously) need card to connect your box to the pstn. Do you have something setup with freeradius already? If not, you could quite easily setup something like this with ASTCC. I have a callback script @ www.aleph-com.net/astpp. Somewhere there. It is way more complicated than you need but you can cut out all the user interaction stuff. Darren Wiebe [EMAIL PROTECTED] Abdul Lateef wrote: Hi friends, I am new in asterisk, i came for CallBack purpose, i read from Voip-info.org aboue callback with asterisk and i am near to collect all information about to start developing callback system. Just i have a samall question, Is Callback needs some special hardware? i have my PSTN phone number i want to call this number after two ring the call will be disconnect and the Callback will start to call back to the caller ID and it should prompt to enter pin id which will authunticate via freeradius.if the authuntication is valid it will give some beep for dialing the international number. Any kind of suggestion will be hearty appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID strings comprised of %23...
I only have the answer to your last question. From my experience, I would go for arbitrary barf. I don't think you are supposed to get anything if there is not a caller id passed. Darren Dave Grey wrote: Well, I am batting close to zero where responses to my questions are concerned, but I suppose I will just keep swinging. I just set up an account with callpacket.com, and noticed that on incoming calls through this provider the values of CALLERID(name) and CALLERID(num) are %23%23%23%23%23%23%23%23%23%23 when the caller has either blocked callerid (tested with *67), or, apparently, sent values that are unexpected (tested via friend who is, for whatever reason, doing SetCallerID(caller 6398A ) on his outbound calls). I have speak caller ID macro that does a system() call to a script on the local machine, and I have been tinkering with ways of handling the different possible strings in some reasonably intelligent way. My question is -- is %23 the escape for the # character here, as I suspect, and if so, is there a way I can tell asterisk to interpret it as such, or do I need to convert it back on my own? Is the %23%23%23%23%23%23%23%23%23%23 (or ###) any kind of an industry standard string, that evaluates to something sensible on a consumer CID display, or is it just some arbitrary barf that callpacket has chosen to send in those cases? Thanks for any info. lyd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip provider in a box
We don't have a complete package quite yet. I think we have most of what you will need but we do not have support at present yet to accept customers payments. We can do that easily via 3rd party sofware but we can't do it ourselves yet. Anyway, www.aleph-com.net/astpp is the link. Darren Wiebe [EMAIL PROTECTED] trixter aka Bret McDanel wrote: I am tasked with evaluating ready made solutions for a voip provider. Does anyone have any recommendations for software, specifically the environment will be a chargable voip provider (ie broadvoice, vonage, etc). They wanted me to see what was there and write something if nothing they like exists. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how many oh323
I've been thinking of using yate http://yate.null.ro/pmwiki/index.php/Main/H323ToSIPSignallingProxy to do this. Any thoughts or experiences? Darren Wiebe [EMAIL PROTECTED] Rob Lith wrote: Altus It's in the transcoding - http://www.voip-info.org/wiki-Asterisk+dimensioning has some notes on oh323 v.s. chan_h323 (chan_h323 is just pass through) - someone says there that you won't be able to run more than *20-25* decent quality calls before asterisk dies when transcoding and H323 are involved. Regards Rob On 10/21/05, *Altus Snyman* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Good day. I configured asterisk and oh323.Im using it as a sip-h323 convertor A call will come in to the asterisk box via IAX and be send to a quintum h323 gateway. in oh323 you can set the max in,out and simultaneous calls, Ive set them all to 100. Calls coming in via iax is alaw and then goes out h323 g729. It is a P4 3.3 and 1Gig of ram.Yet at 20+ calls, calls start failing. Is there someone else with a setup like this.Is the problem on the asterisk side or the quintum Please help Thanks Altus ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Billing
I don't know if astbill supports this or not. ASTPP does supports it though. www.aleph-com.net/astpp You would set admin 1 and admin2 up as resellers. Darren Wiebe [EMAIL PROTECTED] Kanishka Somaratne wrote: Hi I am looking for a asterisk billing system with a reseller module. for example, i there are 2 accoutns admin 1 and admin 2. when they login only the accounts they created should be shown. admin 2s accounts pr rates should not be shown to admin 2. does astbill support this. please let me know regards Kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astcc missing to bill random calls?
What channel are you using to place the calls from ASTCC and what version of asterisk are you using? The get_variable and set_variable perl commands are not working in -HEAD due to stuff being deprecated. Darren Wiebe [EMAIL PROTECTED] maka wrote: Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of hangup. I suppose that astcc.agi was not able to get the answeredime variable from the SIP channel... I have added a few functions to the astcc default script, in order to support different categories of users (functions to get the user type, get the routes and trunks tables for the user category before trytrunk), as well as some 'print SDTERR' statements, in order to trace any problems during execution. Could this be the problem, I noticed that there were reports on the list that get_variable has issues with extensive $agi-verbose callings. I had a problem with get_variable not catching answeredtime once before, and solved these by adding an additional agi-get_variable statement just underneath the first one. Here's how the calls is logged in the csv file: ,38607612,0016318674103,from-sip,38607612 38607612,SIP/sip.mytel.net-0816afc8,,Hangup,,2005-10-17 18:00:16,2005-10-17 18:00:16,2005-10-17 18:00:16,0,0,ANSWERED,DOCUMENTATION The strangest thing is that this appears to happen at random times, so I can't just sit down and watch it through. I would appreciate any ideas, cheers... maka -- I'm sick and tired of being sick and tired... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astcc missing to bill random calls?
Thanks for your feedback. maka wrote: I'm using asterisk-1.0.6. The channel to dial is either SIP or IAX, I've had one missed call in both cases. I commented out the $agi-verbose stuff in many places in the script, and I limited my own print STDERR statements. I haven't seen the isue reappear since then, but I'm not sure whether excessive $agi-verbose output is what caused it. Ok, just wondered. I have also changed the way calls are billled in the calccost function to use includedseconds, and the billing increment period after that. I don't think this has anything to do with the problem anyway.. Wasn't this fixed a while ago? I had a patch that I thought had been accepted.. Darren Wiebe [EMAIL PROTECTED] On 10/19/05, *Darren Wiebe* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: What channel are you using to place the calls from ASTCC and what version of asterisk are you using? The get_variable and set_variable perl commands are not working in -HEAD due to stuff being deprecated. Darren Wiebe [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] maka wrote: Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of hangup. I suppose that astcc.agi was not able to get the answeredime variable from the SIP channel... I have added a few functions to the astcc default script, in order to support different categories of users (functions to get the user type, get the routes and trunks tables for the user category before trytrunk), as well as some 'print SDTERR' statements, in order to trace any problems during execution. Could this be the problem, I noticed that there were reports on the list that get_variable has issues with extensive $agi-verbose callings. I had a problem with get_variable not catching answeredtime once before, and solved these by adding an additional agi-get_variable statement just underneath the first one. Here's how the calls is logged in the csv file: ,38607612,0016318674103,from-sip,38607612 38607612,SIP/sip.mytel.net-0816afc8,,Hangup,,2005-10-17 18:00:16,2005-10-17 18:00:16,2005-10-17 18:00:16,0,0,ANSWERED,DOCUMENTATION The strangest thing is that this appears to happen at random times, so I can't just sit down and watch it through. I would appreciate any ideas, cheers... maka -- I'm sick and tired of being sick and tired... ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- I'm sick and tired of being sick and tired... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC -- semantic note of 'callstart' in cdrs?
This will actually be easy to fix. I'll post a patch along with
someother stuff shortly.
Darren
Darren Wiebe wrote:
That is true. It's just one of those things that is easier to leave
alone to avoid breakage in upgrades. It would be nice to get fixed
though
Darren Wiebe
[EMAIL PROTECTED]
Eric Lyons wrote:
Looking at the code, it would appear that the 'callstart' column of
the cdrs table should really be called 'callend':
$dialstr = IAX2/$res-{path}/$phone|30|HL( .
($maxtime * 60 * 1000) . :6:3);
$res = $AGI-exec(DIAL $dialstr);
$answeredtime = $AGI-get_variable(ANSWEREDTIME);
$dialstatus = $AGI-get_variable(DIALSTATUS);
$callstart = localtime();
return $dialstatus;
No?
Eric.
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Re: [Asterisk-Users] DTMF detection
We had this problem a few months ago but they resolved it for us. I really don't remember more than that. Darren Wiebe [EMAIL PROTECTED] Tom Vile wrote: I have been battling this problem for 2 months with no resolution as of yet with TelaSIP. I am told that it is a provider problem(Level 3) because all TelaSIP is doing is passing the call directly to them once the call comes through. Anyone else having this issue with TelaSIP or Level3? On 10/10/05, *John Millican* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello all, yes there is a lot of information about this on the wiki and in past posts on this list but have not found anything that has solved my problem. setup is: phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf is inband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only. I have only access to dial up so can not go VoIP out of the house. In extensions.conf on colo * i have some logic that based on callerid lets me hit a single digit to get to DISA, this work every time. the problem is that when i enter a number for DISA to dial i get duplicate digits, example i enter 6037862111 and disa tries to dial 6003778621. I have tried setting relaxdtmf=yes in sip.conf with no luck. I have read on the wiki that RFC2833 should work, but alas its a no go. I am also using ulaw which should not be distorting the dtmf through compresion, correct? Also with RFC2833 it should not matter? Everything works great otherwise. sip.conf for colo * is posted below: [general] context=telasip port=5060 bindaddr=0.0.0.0 http://0.0.0.0 srvlookup=yes disallow=all; First disallow all codecs allow=ulaw register = username:[EMAIL PROTECTED] mailto:username:[EMAIL PROTECTED] [telasip] type=peer username=* fromuser=* authname=* secret=* host=gw3.telasip.com http://gw3.telasip.com context=default dtmfmode=RFC2833 disallow=all allow=ulaw canreinvite=no nat=no Thanks in advance for any help John Millican ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC -- semantic note of 'callstart' in cdrs?
That is true. It's just one of those things that is easier to leave
alone to avoid breakage in upgrades. It would be nice to get fixed
though
Darren Wiebe
[EMAIL PROTECTED]
Eric Lyons wrote:
Looking at the code, it would appear that the 'callstart' column of
the cdrs table should really be called 'callend':
$dialstr = IAX2/$res-{path}/$phone|30|HL( .
($maxtime * 60 * 1000) . :6:3);
$res = $AGI-exec(DIAL $dialstr);
$answeredtime = $AGI-get_variable(ANSWEREDTIME);
$dialstatus = $AGI-get_variable(DIALSTATUS);
$callstart = localtime();
return $dialstatus;
No?
Eric.
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Re: [Asterisk-Users] Billing: amaflags and accountcode
The way I do it is to make a list of internal extensions and set those to no charge. They get billed at no charge that way and it works fine. /Plug Starts/ This is done using ASTPP www.aleph-com.net/astpp/ /Plug Ends/ Darren Wiebe [EMAIL PROTECTED] Chris Bagnall wrote: Hi all, I have about 10 SIP phones for different users defined in sip.conf, each with their own accountcode= entry. There is a global setting in sip.conf that states amaflags=documentation There are 3 IAX-PSTN gateways defined in iax.conf for outbound calls. These do not have an accountcode=, but do have amaflags=billing defined in each. The theory was that all calls should be logged, those calls either incoming or between SIP users should have amaflags=documentation (which they do, all well and good), but when a user makes an outgoing call via an IAX gateway, it gets amaflags=billing (so I know it's a chargeable call). However, this doesn't seem to work - all call logs, even those to the IAX gateways all have amaflags=documentation. Is there another way around this? How are you good people using amaflags and accountcode to apportion billing to different users, whilst not billing them for incoming calls or calls between SIP users? Thanks in advance. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC - INUSE Flag
Thanks. I have a question for the mailing list in general. Where should the card get marked as in use? Should it be as soon as you enter the number or should it be when it dials? I don't know for sure. Darren Wiebe [EMAIL PROTECTED] Michael K. Rodriguez wrote: This is my debug with the same issue The agi terminates during the sub tell_time() and exits without calling sub setinuse() or completing the reset of the script. AGI Tx agi_request: astcc.agi AGI Tx agi_channel: Zap/49-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1128401550.162 AGI Tx agi_callerid: xx AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 3 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 33 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: xx AGI Tx agi_rdnis: unknown AGI Tx agi_context: default AGI Tx agi_extension: xx AGI Tx agi_priority: 103 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: xxx AGI Tx 0-r1*CLI AGI Rx ANSWERLI AGI Tx 200 result=0 AGI Rx GET DATA astcc-enter-card-num 6000 -- Playing 'astcc-enter-card-num' (language 'en') AGI Tx 200 result=3546 AGI Rx STREAM FILE astcc-youhave 0123456789 AGI Tx 200 result=0 endpos=4480 AGI Rx SAY NUMBER 11 0123456789 -- Playing 'digits/11' (language 'en') AGI Tx 200 result=0 AGI Rx STREAM FILE astcc-dollars 0123456789 AGI Tx 200 result=0 endpos=6720 AGI Rx STREAM FILE astcc-and 0123456789 AGI Tx 200 result=0 endpos=3680 AGI Rx SAY NUMBER 88 0123456789 -- Playing 'digits/80' (language 'en') -- Channel 0/1, span 3 got hangup request AGI Tx 200 result=-1 == Spawn extension (default, x, 103) exited non-zero on 'Zap/49-1' -- Hungup 'Zap/49-1' -Michael On 10/3/05 10:52 PM, Darren Wiebe [EMAIL PROTECTED] wrote: Can you please post the output with debug agi on ? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in my astcc-exten.conf [incoming] exten = s,1,Answer ;exten = s,2,DeadAGI(astcc.agi) exten = s,2,AGI(astcc.agi) exten = s,3,Hangup I did some Google search on this issue and saw someone else had a problem but no response. -Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC - INUSE Flag
Any developers out there that would like to look at this one? It works fine on Asterisk CVS-v1-0-09/22/05-22:23:34 on a i686 running Linux but it does not work on the 1.2 betas. I agree that the number should be set aside then. I wonder what the problem is. Darren JP Carballo wrote: As soon as the number is entered. Consider the scenario where two people dial in with the same card number. Once one person has entered a valid number, you want to let the other party know as soon as possible that the account is in use. If the card isn't actually in use, it's still best to notify the caller at the earliest. This shouldn't happen of course. Darren Wiebe wrote: Thanks. I have a question for the mailing list in general. Where should the card get marked as in use? Should it be as soon as you enter the number or should it be when it dials? I don't know for sure. snip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC - INUSE Flag
Edit astcc.agi and stick these lines in before sub load_config.
$SIG{HUP} = 'ignore_hup';
sub ignore_hup {
print STDERR \nHUP received!\n\n;
}
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
How do you you apply the patch?
-Scott
- Original Message - From: Nicolás Gudiño [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 05, 2005 9:31 PM
Subject: Re: [Asterisk-Users] ASTCC - INUSE Flag
On 10/5/05, Darren Wiebe [EMAIL PROTECTED] wrote:
Any developers out there that would like to look at this one? It works
fine on Asterisk CVS-v1-0-09/22/05-22:23:34 on a i686 running Linux but
it does not work on the 1.2 betas. I agree that the number should be
set aside then. I wonder what the problem is.
http://bugs.digium.com/view.php?id=5400
Seems to fix the problem... please test and give feedback.
--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] ASTCC - INUSE Flag
Can you please post the output with debug agi on ? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in my astcc-exten.conf [incoming] exten = s,1,Answer ;exten = s,2,DeadAGI(astcc.agi) exten = s,2,AGI(astcc.agi) exten = s,3,Hangup I did some Google search on this issue and saw someone else had a problem but no response. -Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12
I fought with this one for hours last night. I have to get it yet but I'm not sure what the problem is. The permissions are all fine. Any comments anyone? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I just installed the CVS 9-22 and am trying to get ASTCC up and running. I was able to get the web interface config running and it made the database but when I go to the brands page it says there is a problem with the table. Also when I save the config file through the intraface it wont save it to any location. I want to set up a small CC application so if there is a better product to use please let me know. Thanks, Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12
Okay, after spending 12 hours on it I checked the thing that has bit me before. Turn SElinux off. OUCH!! :-) Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I fought with this one for hours last night. I have to get it yet but I'm not sure what the problem is. The permissions are all fine. Any comments anyone? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I just installed the CVS 9-22 and am trying to get ASTCC up and running. I was able to get the web interface config running and it made the database but when I go to the brands page it says there is a problem with the table. Also when I save the config file through the intraface it wont save it to any location. I want to set up a small CC application so if there is a better product to use please let me know. Thanks, Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR problem
Could you post an example of you cdr output. The ASTPP question would be better put on astpp-users. Visit http://aleph.aleph-com.net/mailman/listinfo/astpp-users to subscribe. Darren Wiebe [EMAIL PROTECTED] FaberK wrote: Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel, = Kind of solution...
I will look into this and post back what I find. Darren Wiebe [EMAIL PROTECTED] Ricardo Poppi wrote: Yes Darren. The problem is the same using Zap or SIP. I had no oportunity to verify that using IAX or E1/T1. Rgds, Ricardo Poppi. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recently reported ASTCC audio issues
I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by [EMAIL PROTECTED] on a i686 running Linux. I just spent some time in testing this. I tested the local and IAX2 trunks. Both worked flawlessly. Any comments? Darren Wiebe [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recently reported ASTCC audio issues
I just tested this on Asterisk CVS-v1-0-09/22/05-22:23:34 built by [EMAIL PROTECTED] on a i686 running Linux and it still works perfectly. Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by [EMAIL PROTECTED] on a i686 running Linux. I just spent some time in testing this. I tested the local and IAX2 trunks. Both worked flawlessly. Any comments? Darren Wiebe [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel = Kind of solution...
Have you done any testing to see if it made any difference what type of
trunk was being used?
Darren Wiebe
[EMAIL PROTECTED]
Ricardo Poppi wrote:
Hi all.
I´ve found a kind of solution (if we can call it this way...) and Im
reporting it here to help save some lives.
Editing into astcc.cgi I found where the parameters that set 60 and 30
seconds warning were and put zeros in its place. The last two
lots-of-zeros numbers at second line. So the zap trunk code of astcc.cgi
became like that:
==
if ($res-{tech} eq Zap) {
$dialstr = Zap/$res-{path}/$phone|30|HL( . ($maxtime *
60 * 1000) . :0:0);
$res = $AGI-exec(DIAL $dialstr);
$answeredtime = $AGI-get_variable(ANSWEREDTIME);
$dialstatus = $AGI-get_variable(DIALSTATUS);
$callstart = localtime();
return $dialstatus;
}
==
And - at least until now... - everything is working fine. The credit is
being take from the cards in the right amount and no warnings are being
given when 60 and 30 seconds left. When credit finishes, the agi script
just finishes the call.
If somebody has a better way to do that, please let us know.
Rgs, Ricardo Poppi.
Mensagem Original
Assunto: ASTCC speaks and cut RTP channel
Data: Fri, 09 Sep 2005 18:09:52 -0300
De: Ricardo Poppi [EMAIL PROTECTED]
Para: asterisk-users@lists.digium.com
Hi list.
I have a fine running Ser+Asterisk environment and have just installed
ASTCC. It´s working fine either, including its caller-id authentication
feature (the one we pass the card-number as CALLERID variable and
number-to-dial as EXTEN variable).
The issue, a great one, is that when the credit is about one minute to
end, the ASTCC prompt gets into the call, says that you have one minute
left... and when it was suppose to leave and let the RTP traffic of the
original call be reestablished, it never happens. The RTP packets - I
could see that at asterisk debug screen - stop running and the call is
still signaled as active, but no media at all.
This is a serious problem I´m having and, as I could see, I´m not the
only one. Mr. Chilini reported that around jun 30th this year, as you
can see bellow: (I just added a comment at this voip-info page to see if
anyone could give some clues about that)
http://www.voip-info.org/tiki-index.php?page=ASTCCGuide#comments
Do anyone here in this list had any situation alike? Do you have any
clues do help me? (and others because it will be documented, of course).
Thanks in advance,
Ricardo Poppi.
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