Re: [asterisk-users] GSM cellphone as cheap gateway?
There is also chan_sebi and chan_celliax. I tried chan_mobile without success (too unstable). Those two channels above are still in my pending list. On Mon, Sep 21, 2009 at 8:54 AM, Vijay Gandhi vi...@gandhiinfotech.comwrote: It is actually FCT, my mistake I wrongly typed in FTC. FCT is Fixed Cellular Terminal, you can put your GSM card into it and it gives you an output of a PSTN line (FXs) which can be connected to your FXo device, normally in india, we get these devices for about $50 (USD Fifty only). Regards Vijay Gandhi GIPL(An ISO 9001:2000 Company) +91-9811688460 +44-2080992384 vi...@gandhiinfotech.com www.gandhiinfotech.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian Victor Sent: Monday, September 21, 2009 6:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] GSM cellphone as cheap gateway? Olivier schrieb: 2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com There are FTC's available, What is it (a FTC) ? a cable ? Any pointer to that (Google is helpless)? ? My guess would be fixed to cell or FX to cell adapter. Chris ___ -- 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 ___ -- 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 ___ -- 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
[asterisk-users] chan_mobile future
Hello list, I wonder what are the ideas to improve chan_mobile implementation to make it usb compliant. I am highly interested as a developer and I want to know if there is any initial project. I read part of the chan_mobile source code and a TS of hands-profile-protocol trying to understand better the environment. After that, I started some research about the possibility of doing the same comminication style using standard AT commands but apparently there is no way to redirect audio stream between a remote host and the phone using serial comm. device like USB cable. Finally, I thought about writing a middleware for Symbian C++ phones that could act as call manager and redirect the audio to Linux box using serial connection. Here I stopped because I don't find any valid example of voice streaming, only plain data. Besides, I am not sure if this is possible. A little help to make Asterisk better is always useful :) Regards. ___ -- 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
[asterisk-users] Modem
Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. ___ -- 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] Modem
I did not know that the price was that low. Anyway, for people living really far from USA the price gets incremented twice or more and this is without considering the conversion between currencies. 1 $ = 5100 Gs., not cheap at all. Thanks. On Sun, Aug 2, 2009 at 3:07 PM, jon pounder j...@inline.net wrote: Carlos Ruiz Diaz wrote: Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. the digium single fxo cards and clones for about $10 ARE modems. you can get a sip gateway fxo + fxs in one box for about $50 really - how much cheaper do you want ? ___ -- 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 ___ -- 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 ___ -- 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] Modem
You understand perfectly fine the situation :) . I'm not saying that Paraguay has the worse economy in South-America, but we need to work much harder to get latest technology or to mount a tiny/small laboratory. You will get amized if you see the things that we have done with pieces of hardware considered as garbage in USA :D Does Antelco still dominate the telco market in Paraguay, I wonder. Yes, they changed their name to Copaco for Compania Paraguaya de Comunicaciones. It's basically the same company ruling the whole country. :S Thanks to all for answering my question. On Sun, Aug 2, 2009 at 4:56 PM, Jared Smith jsm...@digium.com wrote: On Sun, 2009-08-02 at 14:54 -0400, Carlos Ruiz Diaz wrote: Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. This question has been answered many times, but just for the fun of it I'll answer it again: If PC modems had been ideal telephony cards, we'd still be using them. My own experience with using modems as FXO devices (long before I became a Digium employee) was that they were awful. I encountered problems with echo, half-duplex audio, and lack of far-end disconnect supervision. All of those problems are solved with most modern telelphony cards (except for the ultra-cheap cards, which are still just modems). To put it frankly, I wouldn't wish one of those modems on my worst enemies. Anyway, for people living really far from USA the price gets incremented twice or more and this is without considering the conversion between currencies. 1 $ = 5100 Gs., not cheap at all. I understand that the cards are disproportionately expensive in many parts of the world as compared to the United States, because of the difference in economies. I spent a couple of years in Paraguay in the mid 90s, and know what it's like to pay outrageous prices for specialized electronics just because they have to be imported from other countries. (I'm guessing that you're from Paraguay, based on on the monetary conversion you gave. Does Antelco still dominate the telco market in Paraguay, I wonder?) That being said, the cost per port of the Digium cards (or any of our competitors who design their own cards) is still much lower than what you'd pay for traditional telephony cards, such as those manufactured by Dialogic or Aculab. I know that probably doesn't help you afford to be able to buy a more expensive card, but hopefully you have a better understanding of why we don't use modems as FXO devices. If your time and sanity are worth anything at all, it's a worthwhile investment to buy a good solid telephony card. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 ___ -- 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] Latest chan_mobile
That is exactly what happens to me. Still looking for a solution. On Wed, Jul 22, 2009 at 9:44 AM, Thomas Kenyon dig...@sanguinarius.co.ukwrote: Carlos Ruiz Diaz wrote: @Steve: I considered the hardware separation between servers but when I exposed the idea it was immediately discarded because it is mandatory to have all in a box. Well, I'll start the migration then. Thank you. I doubt this helps anyone, but today I built the newest stable kernel (2.6.30.2) and the latest bluez libs (bluez-4.46) and obviously rebuilt dahdi and asterisk-addons. Without any config changes chan_mobile is working for incoming calls, picking up the handset is answeing the calls, and there is 2 way audio (which wasn't working before). Oddly when a call finishes, the mobile disconnects for a while and then reconnects again and there is terrible audio with outgoing calls, (scratchy and with a few seconds delay). This is definite progress (and doesn't require a separate box). This is all with a Cambridge Silicon Radio USB2 dongle and a nokia e61. ___ -- 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] Latest chan_mobile
Yes, I tried with: Dell Computer Corp. Wireless 355 Bluetooth, built-in Encore, USB adapter. Always with: Nokia N80 Kernel: 2.6.27.21-0.1-pae. On Wed, Jul 22, 2009 at 10:46 AM, Thomas Kenyon dig...@sanguinarius.co.ukwrote: Carlos Ruiz Diaz wrote: That is exactly what happens to me. Still looking for a solution. Well, it's a step forward from what I was getting before. Have you tried with different USB adapters and handsets? ___ -- 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] Latest chan_mobile
@Steve: I considered the hardware separation between servers but when I exposed the idea it was immediately discarded because it is mandatory to have all in a box. Well, I'll start the migration then. Thank you. On Sun, Jul 19, 2009 at 12:59 AM, Steve Totaro stot...@totarotechnologies.com wrote: No need to migrate, just have a chan_mobile server to hand the calls over via SIP. It is your cell phone network gateway I like to separate functions to different boxen. Database on one, Asterisk on another, TDM - SIP gateway on another, GUI/CRM somewhere else. Why not have a Cell - SIP gateway? Just my approach but it seems to work well. Power and RU space aside. Thanks, Steve Totaro On Sat, Jul 18, 2009 at 11:23 PM, Sasa Bobek sasa.bobek...@gmail.comwrote: Yes, chan_mobile works great on Elastix. If the migration is complicated, you may consider installing/testing it on an old computer. On Sun, Jul 19, 2009 at 2:21 AM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank for your time. Do you used chan_mobile with Elastix distribution successfully? If so, I will consider the switch. I can't jump to another distribution easily because I have a working environment that will make really hard the migration. On Sat, Jul 18, 2009 at 10:57 AM, Sasa Bobek sasa.bobek...@gmail.comwrote: In general, I found it hard to get chan_mobile working straight out of the box, and although there is a great effort to make it so, phone manufacturers are not helping by making command sets and BT implementations different from device to device, SW version to SW version. Elastix seems to have the most trouble free implementation out there and has certainly saved me a lot of time and money and I recommend you give it a go, before banging your head over code. You can check the buglist on Digium for further info or the list of compatible phones on voip-info.org, but it may be a USB dongle issue as well (CSR seems to be the safest bet after they fixed the error log flood). On Sat, Jul 18, 2009 at 3:27 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Hello I recently updated my asterisk-addons-1.6.2 to the last revision and I have this problem that I don't know how to interpret, bug or not. I connected a Nokia N80 phone to use chan_mobile and everything works great until the phone starts getting disconnected after the call finished and sometimes during the call attempt. Is this a bug or a possible known issue for Nokia phones? # rpm -qa | grep blue pulseaudio-module-bluetooth-0.9.12-10.1 bluez-utils-3.36-7.1 kdebluetooth4-0.3-4.1.1 libbluetooth-devel-3.36-3.1 gnome-bluetooth-0.11.0-26.2 bluez-test-4.22-6.1.1 libbluetooth3-4.22-6.1.1 libbluetooth2-3.36-3.1 Thanks in advance! Carlos. ___ -- 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 -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Latest chan_mobile
Hello, I recently updated my asterisk-addons-1.6.2 to the last revision and I have this problem that I don't know how to interpret, bug or not. I connected a Nokia N80 phone to use chan_mobile and everything works great until the phone starts getting disconnected after the call finished and sometimes during the call attempt. Is this a bug or a possible known issue for Nokia phones? # rpm -qa | grep blue pulseaudio-module-bluetooth-0.9.12-10.1 bluez-utils-3.36-7.1 kdebluetooth4-0.3-4.1.1 libbluetooth-devel-3.36-3.1 gnome-bluetooth-0.11.0-26.2 bluez-test-4.22-6.1.1 libbluetooth3-4.22-6.1.1 libbluetooth2-3.36-3.1 Thanks in advance! Carlos. ___ -- 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] Latest chan_mobile
Thank for your time. Do you used chan_mobile with Elastix distribution successfully? If so, I will consider the switch. I can't jump to another distribution easily because I have a working environment that will make really hard the migration. On Sat, Jul 18, 2009 at 10:57 AM, Sasa Bobek sasa.bobek...@gmail.comwrote: In general, I found it hard to get chan_mobile working straight out of the box, and although there is a great effort to make it so, phone manufacturers are not helping by making command sets and BT implementations different from device to device, SW version to SW version. Elastix seems to have the most trouble free implementation out there and has certainly saved me a lot of time and money and I recommend you give it a go, before banging your head over code. You can check the buglist on Digium for further info or the list of compatible phones on voip-info.org, but it may be a USB dongle issue as well (CSR seems to be the safest bet after they fixed the error log flood). On Sat, Jul 18, 2009 at 3:27 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Hello I recently updated my asterisk-addons-1.6.2 to the last revision and I have this problem that I don't know how to interpret, bug or not. I connected a Nokia N80 phone to use chan_mobile and everything works great until the phone starts getting disconnected after the call finished and sometimes during the call attempt. Is this a bug or a possible known issue for Nokia phones? # rpm -qa | grep blue pulseaudio-module-bluetooth-0.9.12-10.1 bluez-utils-3.36-7.1 kdebluetooth4-0.3-4.1.1 libbluetooth-devel-3.36-3.1 gnome-bluetooth-0.11.0-26.2 bluez-test-4.22-6.1.1 libbluetooth3-4.22-6.1.1 libbluetooth2-3.36-3.1 Thanks in advance! Carlos. ___ -- 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] Using a mobile phone via USB as an extension
I read it in this list. I buit an application on top of chan_mobile and i needed usb connectivity to improve the bandwidth so i googled for the answer and one of the hits was from here. On 7/9/09, Olivier oza-4...@myamail.com wrote: 2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. Where did you get this info (USB connectivity for chan_mobile) ? Is there a way to learn a bit more ? -- Sent from Gmail for mobile | mobile.google.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Variable using AMI
Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. Thanks in advance. ___ -- 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] chan_mobile help.
Try upgrading your bluez library. You can also try a parallel installation with the last revision of chan_mobile. Use the same phone always to discard any phone issues. On Mon, Jul 6, 2009 at 11:43 AM, Razza razz...@gmail.com wrote: Thanks for your response. I gave loads of info in my original mail, surely someone can help without jumping distro? ___ -- 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
[asterisk-users] Get channel string
Hello, When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique ID but I don't know how to retrieve it immediately after the originate action to be able to use it to identify the calls that I made. How can I get the actual channel string after calling Originate? or how can I get the unique ID of a call about to start (or already started) using the same action (Originate). Regards. Carlos. ___ -- 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] Variable using AMI
Thank you! I did not know the existence of DB command. The command allows me to store KVPs but I have to use the same variable name every time so every process that starts the AMI instance will override the values making it unusable for what I want to achieve. It was really useful anyways. :) 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com carlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- 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] Get channel string
No :( . Response gave me an empty unique-Id. Apparently it is generated on the fly once the resources are allocated or something else. I don't have any channel information in the response. On Mon, Jul 6, 2009 at 3:22 PM, Philipp Kempgen philipp.kemp...@amooma.dewrote: Carlos Ruiz Diaz schrieb: When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique ID but I don't know how to retrieve it immediately after the originate action to be able to use it to identify the calls that I made. How can I get the actual channel string after calling Originate? or how can I get the unique ID of a call about to start (or already started) using the same action (Originate). Doesn't the OriginateResponse give you that information? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Variable using AMI
I am sorry for my bad English. Apparently I'm explain myself wrongly but you got the point. I tried GetVar as AMI action but I have to specify a channel string. Of course I have the channel string, I parametrized it but Asterisk adds another string to the original channel and I can't obtain the variable value because of the lack of the real channel string. eg. I write SIP/1000 in the channel param. but asterisk adds SIP/1000-*12eg12*. Obviously is not always the same string. This is just a illustrative demonstration. I wrote a mail asking for help with that problem. Thanks. 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com Well, I do not understand very well what you are trying to do, but I'll give you some advice: If you want a variable only for the AGI you call, you just have to declare that variable on the AGI. If you would like to make visible that variable as long as the call is active and for each call, even if the name is the same, you have to set a channel variable with the Set(variable=value) command. If you would like to have a variable shared between two or more channels, use the SHARED() funcion(Asterisk 1.6, back ported to 1.4) If you want a variable to be accessed from all the channels, you could use a global variable. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, Juan Carlos Ruiz Diaz wrote: Thank you! I did not know the existence of DB command. The command allows me to store KVPs but I have to use the same variable name every time so every process that starts the AMI instance will override the values making it unusable for what I want to achieve. It was really useful anyways. :) 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com carlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- 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 ___ -- 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] Dial
Thank you for your help.
Apparently I will rollback to AMI interface. I found a project named
Asterisk.NET that interface AMI and my mono-C# application. I will be
working in a solution.
BTW, It will be truly interesting the possibility of writing dialplans in C#
having all of the advantages that managed code and a matured framework (Mono
project) can give us as developers.
On Fri, Jul 3, 2009 at 3:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jul 02, 2009 at 09:02:59PM -0700, Steve Edwards wrote:
Access through the CLI (via -r -x) is problematic because you are subject
to the whims of another process changing verbosity levels and you may
have
to sift through a lot of cruft as well as the overhead of invoking a
process for each command.
I believe that this has changed in recent versions (recent 1.4 and
1.6.x).
Access through ${astrundir}/asterisk.ctl would be interesting, especially
if more than 1 process attempts access.
Each of them gets its own socket. Just like multiple 'asterisk -r'
connections. Use either socat or OpenBSD netcat (both support writing to
a unix-domain socket). This should work reasonably well if you just want
to write some commands but are not interested in their output.
--
Tzafrir Cohen
icq#16849755
jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Dial
That is the idea :) . I'll search the web for more info. On Fri, Jul 3, 2009 at 9:09 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Fri, Jul 03, 2009 at 08:49:41AM -0400, Carlos Ruiz Diaz wrote: Thank you for your help. Apparently I will rollback to AMI interface. I found a project named Asterisk.NET that interface AMI and my mono-C# application. I will be working in a solution. BTW, It will be truly interesting the possibility of writing dialplans in C# having all of the advantages that managed code and a matured framework (Mono project) can give us as developers. This will require the Asterisk process to have some .Net interpreter. This is not undoable (either mono or Portable .Net could be used), but still requires some actual work. Python seems to be more popular among the target audince and I still don't see such a pbx_python. If you really want it, have someone write it :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Using a mobile phone via USB as an extension
Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill t...@nickhill.co.uk wrote: I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is the same as a land line. In other countries, it is very different. I know of a mobile operator who bundle lots of free on-network minutes with SIM cards. I wonder if it is possible to forward the call via a mobile phone tethered to an asterisk server through USB? Has anyone tried tethering a mobile phone to an asterisk server and configuring it as an asterisk extension so they can use free or cheap on-network minutes for the mobile leg of the call? Thanks. Nick. ___ -- 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
[asterisk-users] Dial
Hello list, I want to know the options that I have as an Asterisk user to make a call and control its progress. I have been using the spool directory to originate my calls using this format: Channel: Callerid: MaxRetries: RetryTime: WaitTime: Context: Extension: Priority: Is there any other way to originate it and being able to cancel it whenever it's necessary. I know I can write the Dial command using the CLI but the point is the automation that a computer program (that I will write) can give me. Thanks in advance. ___ -- 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] Dial
Thank you Steve. AMI is what I used before and now I am looking for something better if exists. On Thu, Jul 2, 2009 at 8:20 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote: I want to know the options that I have as an Asterisk user to make a call and control its progress. I have been using the spool directory to originate my calls using this format: Is there any other way to originate it and being able to cancel it whenever it's necessary. I know I can write the Dial command using the CLI but the point is the automation that a computer program (that I will write) can give me. Visit voip-info.org and read up on AMI. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Nagios Asterisk
*FAILS= UP*
should be
*FAILS=UP*
(without the space)
it is a syntax error and if you test the script in console you will notice
it immediately.
On Wed, Jun 17, 2009 at 2:34 PM, Sriram d_r_sri...@hotmail.com wrote:
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to show
them line by line Up or down) using nagios using below script, but i always
get the status as down and red..can anyone let me know how to read an output
from nagios plugin ? nagios etc is configured already and is working
PATH=/bin:/sbin:/usr/bin:/usr/sbin
FAILS=
SPANS=$(asterisk -rnx pri show span
1 | grep -a PRI | awk '{print $3;}' | cut -d/ -f1)
STATUS=$(asterisk -rnx pri show span 1 | grep -a Status | awk
'{print $3;}' | cut -d, -f1)
if [ $STATUS == Up ]; then
FAILS= UP
else
FAILS=Down
fi
if [ $FAILS != UP ]; then
echo ISDN Lines down
exit 2
fi
echo ISDN OK
exit 0
If anyone can share the above script for asterisk monitoring then i wud be
grateful
rgds
Sriram
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[asterisk-users] Writing for asterisk
Hello, Where can I found information about writing modules, applications and low level interactions for asterisk? At http://www.asterisk.org/developers I was unable to find tutorials for doing what I mentioned above. Thanks in advance. Carlos. ___ -- 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] Writing for asterisk
Thank you! It is really useful, On Thu, Jun 11, 2009 at 9:13 PM, Moises Silva moises.si...@gmail.comwrote: this is possibly the best you can find: http://www.russellbryant.net/blog/2008/06/30/how-to-write-an-asterisk-module-part-3/ On Thu, Jun 11, 2009 at 7:50 PM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com wrote: Hello, Where can I found information about writing modules, applications and low level interactions for asterisk? At http://www.asterisk.org/developers I was unable to find tutorials for doing what I mentioned above. Thanks in advance. Carlos. ___ -- 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialer program
Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my time in something that nobody is going to use. Do you people think that a dialer could be considered a successful project? Thanks in advance. Carlos ___ -- 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] Dialer program
Thank you Jose. Interesting suggestion! Is there any other? On Wed, Jun 10, 2009 at 11:21 AM, Jose P. Espinal j...@slackware-es.comwrote: Hola Carlos, Have you searched for ViciDialer? It's a good one. Give it a shot, it might be what you are looking for. Carlos Ruiz Diaz wrote: Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my time in something that nobody is going to use. Do you people think that a dialer could be considered a successful project? Thanks in advance. Carlos ___ -- 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 -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs ___ -- 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] Dialer program
I can't find GNUDial web page :( On Wed, Jun 10, 2009 at 1:44 PM, Jaswinder Singh vick...@gmail.com wrote: There is also GNUdial but i would prefer VICIdial anyday over it ( personal opinion :) ) . On Wed, Jun 10, 2009 at 9:43 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you Jose. Interesting suggestion! Is there any other? On Wed, Jun 10, 2009 at 11:21 AM, Jose P. Espinal j...@slackware-es.comwrote: Hola Carlos, Have you searched for ViciDialer? It's a good one. Give it a shot, it might be what you are looking for. Carlos Ruiz Diaz wrote: Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my time in something that nobody is going to use. Do you people think that a dialer could be considered a successful project? Thanks in advance. Carlos ___ -- 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 -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs ___ -- 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] Open source SIP client
I also have problems writing in English and I use to ignore this kind of treatment but your discriminative reaction just makes me sick and I feel sorry for you. English is not a hard language to learn/understand and it is probably perfect to be the most used language (after Chinese :S) in the earth and you should not feel superior for speaking it. If there is any syntax/semantic error in my text, don't bother trying to correct it. If you have enough IQ it will be understandable for you. On Mon, May 18, 2009 at 8:07 PM, Alex Balashov abalas...@evaristesys.comwrote: Pascal Bruno wrote: intelligence to understand what somebody who's english is not the primary language wanted to say and put some effort to guide or help someone in the community getting to the right direction instead of trying to put him down. If language were the only problem, there would be no hostile reception to his question. Non-native English speakers are ubiquitous in this space. I think the more discerning members of the audience recognise that there are deeper issues at work here related to the mental disposition from which the line of questioning is proceeding. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] Open source SIP client
I think you misunderstood my statement. I am not referring to any aspect of language mechanics or language usage whatsoever Have you considered investigating more deeply the basic mechanics of the written English language? Or do you simply have not the foggiest clue what these collections of syllables intended to convey meaning -- words I don't think so. do vs. think aspect Yes, I have to admit that you are right saying that. When I don't know how to write/read something, I just http://translate.google.com.py/translate_t?hl=es# On Mon, May 18, 2009 at 8:42 PM, Alex Balashov abalas...@evaristesys.comwrote: I think you misunderstood my statement. I am not referring to any aspect of language mechanics or language usage whatsoever, but rather to the do vs. think aspect of the inquiry, which can be posed in a range of ways -- from the most refined and polished English to the most broken imaginable. Forget English. Nothing to do with English. Carlos Ruiz Diaz wrote: I also have problems writing in English and I use to ignore this kind of treatment but your discriminative reaction just makes me sick and I feel sorry for you. English is not a hard language to learn/understand and it is probably perfect to be the most used language (after Chinese :S) in the earth and you should not feel superior for speaking it. If there is any syntax/semantic error in my text, don't bother trying to correct it. If you have enough IQ it will be understandable for you. On Mon, May 18, 2009 at 8:07 PM, Alex Balashov abalas...@evaristesys.commailto: abalas...@evaristesys.com wrote: Pascal Bruno wrote: intelligence to understand what somebody who's english is not the primary language wanted to say and put some effort to guide or help someone in the community getting to the right direction instead of trying to put him down. If language were the only problem, there would be no hostile reception to his question. Non-native English speakers are ubiquitous in this space. I think the more discerning members of the audience recognise that there are deeper issues at work here related to the mental disposition from which the line of questioning is proceeding. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile and DTMF
Hello list, I just updated to the last release my asterisk-addons copy and the DTMF works almost perfect. I have the same situation that I used to have using the stable version 1.6.1. The tones gets detected but only one keypress in a given time. I don't know if this is a configuration problem with my mobile.conf or if is the same bug that was reported previously. I am also receiving error reports during socket read: *[May 15 17:45:12] ERROR[21830]: chan_mobile.c:1038 mbl_read: read error 107* Bellow, my configuration file, perhaps there is something wrong with it. [general] interval=30 [adapter] id=built-in address=00:1D:D9:EB:FA:6F ;forcemaster=yes ;alignmentdetection=yes [NN80] address=00:12:D2:30:DB:FB port=28 context=mobile adapter=built-in group=1 ;nocallsetup=yes dtmfskip=200 dtmfmode=auto I hope you can help me. Carlos. ___ -- 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] chan_mobile and DTMF
Hello! I tested the last revision of chan_mobile.c but I am still getting errors during the call. The last time the error starts when the call was established but with this new revision it starts sooner during the call attempting. The following error is reported: [May 8 19:40:48] ERROR[6971]: chan_mobile.c:1038 mbl_read: read error 107 Regards. Carlos. On Thu, May 7, 2009 at 10:37 AM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you. Please, any patch you think can have possibility to work, just let me know and I will do the tests and post the results. Regards On Thu, May 7, 2009 at 10:33 AM, Matthew Nicholson mnichol...@digium.comwrote: Sorry. I am not sure which patch I wanted you to try now. The issue I posted may be related to your issue. On Wed, 2009-05-06 at 14:53 -0400, Carlos Ruiz Diaz wrote: I think I misunderstood your mail. There is no patch available yet, right? I went to the page you linked but I did not found a patch file. On Wed, May 6, 2009 at 2:45 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you very much! I'll try with the patch and post the results. On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson mnichol...@digium.com wrote: Try the patch on this bug http://bugs.digium.com/view.php?id=15042 I don't get that error with my setup, but others have seen it. I am fairly sure of what is causing it. Still working on a fix. On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote: I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.com wrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine
Re: [asterisk-users] Messaging System
I use an application that fits exactly your needs but is not free. You can ... - load a call file containing the numbers to call. - load the call file from a db. - design you own context using a simple editor that generates an AEL compatible dialplan and manages the existing contexts saved in extension.ael. - load sound files to the server in any compatible format or use its converter to change the format to .gsm. - call and see the progress in real time. - schedule call(s) and configure the start time and end time. When you reach the end time the software will reschedule the unstarted call(s) to the next day or next X hours or will do nothing if you wish. - have billing information. - have CDR entries. The software has two parts. The server runs in a Linux box using mono runtime and the front-end can be executed using Windows, Linux or MacOS X (was made in C# and it's compatible with mono 2.4 or above) On Wed, May 6, 2009 at 11:19 PM, Ricardo Melendez rmelen...@utep.com.mxwrote: 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 ___ -- 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] chan_mobile and DTMF
Thank you. Please, any patch you think can have possibility to work, just let me know and I will do the tests and post the results. Regards On Thu, May 7, 2009 at 10:33 AM, Matthew Nicholson mnichol...@digium.comwrote: Sorry. I am not sure which patch I wanted you to try now. The issue I posted may be related to your issue. On Wed, 2009-05-06 at 14:53 -0400, Carlos Ruiz Diaz wrote: I think I misunderstood your mail. There is no patch available yet, right? I went to the page you linked but I did not found a patch file. On Wed, May 6, 2009 at 2:45 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you very much! I'll try with the patch and post the results. On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson mnichol...@digium.com wrote: Try the patch on this bug http://bugs.digium.com/view.php?id=15042 I don't get that error with my setup, but others have seen it. I am fairly sure of what is causing it. Still working on a fix. On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote: I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.com wrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press
Re: [asterisk-users] chan_mobile and DTMF
Thank you very much! I'll try with the patch and post the results. On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson mnichol...@digium.comwrote: Try the patch on this bug http://bugs.digium.com/view.php?id=15042 I don't get that error with my setup, but others have seen it. I am fairly sure of what is causing it. Still working on a fix. On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote: I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.com wrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- 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 -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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 -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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] chan_mobile and DTMF
I think I misunderstood your mail. There is no patch available yet, right? I went to the page you linked but I did not found a patch file. On Wed, May 6, 2009 at 2:45 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.comwrote: Thank you very much! I'll try with the patch and post the results. On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson mnichol...@digium.comwrote: Try the patch on this bug http://bugs.digium.com/view.php?id=15042 I don't get that error with my setup, but others have seen it. I am fairly sure of what is causing it. Still working on a fix. On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote: I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.com wrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- 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 -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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 -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile and DTMF
Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- 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] chan_mobile and DTMF
Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.comwrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- 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 -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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] chan_mobile and DTMF
I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.comwrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunkasterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.comwrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- 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 -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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
