[asterisk-users] Farewell
I just wanted to wish all of you good luck I'm officially retired and will be removing my name from the list. I can attest that this list has been a great help throughout my career. I have deployed probably over 100 installations over a 10-year period. Any of you newcomers this list the most valuable tool you can have. Sincerely, Vincent MedinaInformation Systems DirectorAPCN, Inc. (305)785-3355 Sent using www.apcn.net Internet Services. Original message From: Dario Estupinan <darioestupi...@soygenial.co> Date: 08/17/2016 8:53 AM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13 REMOVE ME please. 2016-08-15 15:16 GMT-05:00 Jonas Kellens <jonas.kell...@telenet.be>: Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc77@178.19.90.240>' failed for '78.119.140.190:5076' - Wrong password [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc78@178.19.90.240>' failed for '78.119.140.190:5072' - Wrong password [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc79@178.19.90.240>' failed for '78.119.140.190:5062' - Wrong password [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc80@178.19.90.240>' failed for '78.119.140.190:5060' - Wrong password [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc81@178.19.90.240>' failed for '78.119.140.190:5060' - Wrong password Is this a known problem ?? Second question I have : can I get the complete list of columns that can be used in realtime database for sip peers somewhere (update for Ast 13) ? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, dtlssetup possible ?? Thanks for the help. Kind regards. Jonas. -- _ -- 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 -- DARIO ESTUPIÑAN V.Líder de NOC+Cel: 3008832295E-Mail: darioestupi...@soygenial.co Antes de imprimir este mensaje, asegúrese de que es necesario. Proteger el medio ambiente está también en sus manos. AVISO LEGAL: Este mensaje es confidencial, puede contener información privilegiada y no puede ser usado ni divulgado por personas distintas de su destinatario. Si recibe este correo por error, por favor elimínelo y avise a su remitente. Está prohibida su retención, grabación, utilización, aprovechamiento o divulgación con cualquier propósito. La Corporación Politécnica Nacional de Colombia no asume ninguna responsabilidad por eventuales daños generados por el recibo y el uso de este material, siendo responsabilidad del destinatario verificar con sus propios medios la existencia de virus u otros defectos. El presente correo electrónico solo refleja la opinión de su Remitente y no representa necesariamente la opinión oficial de la Corporación o de sus Directivos. -- _ -- 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] PRI got event HDLC Abort
You're HDLC error is evident of timing slips. Use cat /proc/dahdi/1 or 2 or 3 Also cat /proc /interrupts -- Vincent Swart On Mon, Nov 5, 2012 at 8:00 PM, asterisk-users-requ...@lists.digium.comwrote: 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: Asterisk Support from Digium (Danny Dias) 2. Re: Asterisk Support from Digium (Chris Bagnall) 3. Re: PRI got event HDLC Abort (Edwin Lam) 4. Re: PRI got event HDLC Abort (Thorsten G?llner) 5. play wav file (Jerry Geis) 6. Re: play wav file (Danny Nicholas) 7. Re: play wav file (Christopher Harrington) -- Message: 1 Date: Sun, 4 Nov 2012 21:37:27 +0100 From: Danny Dias ing.diasda...@gmail.com Subject: Re: [asterisk-users] Asterisk Support from Digium To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: ca+d0ut_xh_bh3g2mk1k8anqghbcs3tro94cn3f+tlt0ie6j...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Thanks Andrew, But i'm quite confuse with the following: *Q: Does Digium offer SLA guaranteed support for Asterisk?* *A:* Yes. Digium offers SLA guaranteed support, to SLA-entitled customers, for the Certified Asterisk branches. Digium does not offer SLA guaranteed support for other branches or releases. Just for Certify Versions of Asterisk? What does SLA means exactly? For example, if i install a FreePBX/Elastix (i'm not a good friend of these systems, but customers always ask for a web interface for management) to a customer, can i buy support from Digium for the Asterisk Release used? It would be nice to now the scope and limits of this support Thanks 2012/11/3 Andrew Latham lath...@gmail.com On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello, I wonder if Digium provides support for Asterisk OpenSource versions as an anual fee or something? For example, if i download Asterisk 1.8.X (Certified or not...) can i buy support from Digium to maintain and help on possible future problems in my configuration? Thanks Yes Please review http://www.digium.com/en/supportcenter/custom-communications-solutions/ for more information. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.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 -- *SIP:* da...@voice.danntel.net http://www.danntel.net/?page_id=189 *Web: *http://www.danntel.net -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20121104/dacdf299/attachment-0001.htm -- Message: 2 Date: Sun, 04 Nov 2012 22:33:39 + From: Chris Bagnall aster...@lists.minotaur.cc Subject: Re: [asterisk-users] Asterisk Support from Digium To: asterisk-users@lists.digium.com Message-ID: 5096ed43.5060...@lists.minotaur.cc Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 4/11/12 8:37 pm, Danny Dias wrote: For example, if i install a FreePBX/Elastix I'd be very surprised (no, actually, I'd be *amazed*) if Digium were prepared to provide support on a product from a third party, which is what FreePBX and Elastix effectively are. Kind regards, Chris -- This email is made from 100% recycled electrons -- Message: 3 Date: Sun, 04 Nov 2012 21:13:35 -0800 From: Edwin Lam edwin@officegeneral.com Subject: Re: [asterisk-users] PRI got event HDLC Abort To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 50974aff.1010...@officegeneral.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 11/2/2012 10:06 PM, Liban Abdi wrote: is there static on the line?? no. there were customer complains about sound cutting in and out. however i wasn't noticing and bad sound quality when i was testing it. is there timing slips and crc4 errors? no. the only messages i have are the HDLC abort warning
Re: [asterisk-users] PRI got event HDLC Abort
I experienced this exact message this week. I'm sure it has to do with the interface card sharing IRQs. You will see timing slips increment from cat /proc/dahdi/1. Change PCI slot or re-assign an IRQ and this should be fixed. -- Vincent Swart -- _ -- 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] Benchmarking AGI performance in C, PHP, and Perl
On 11/07/11 23:42, Steve Edwards wrote: On 12/07/11 9:29 AM, Steve Edwards wrote: Many times, I've made the statement that you can execute hundreds of AGIs written in C in the time it takes to load an interpreter and parse a script written in PHP or Perl. Well, now that I know better, let's not perpetuate an ancient claim. 'Dozens' is more appropriate with current hardware. On Tue, 12 Jul 2011, Matt Riddell wrote: It would be interesting to see the same types of tests run against fast-agi - personally if I write an agi that will be called 1000 times I'm going to leave it running and have network requests against it rather than starting and stopping every time. Isn't every AGI going to be executed 1,000s of times over it's lifetime? 'Standalone' AGIs still have advantages in lower complexity and less impact on failure. If a bug takes out your fastagi daemon it can affect all calls. I'm pretty sure if you have a bug in your AGI code it's going to affect all calls whether its fastagi or not. Unless the bits between the AGI and DB calls are significant, there should be no significant difference between source languages in a fastagi environment. One of my other objections to scripting languages are that they don't 'catch' stupid errors for me (like a misspelled variable name) as well as compiled languages. 'gcc -Wall' is my friend. Also they tend to be used more by 'non-programmers' who get away with 'stupid' stuff like calling out to system() and piping a bunch of commands together because they don't know how to use the language properly :) Mind if I post it to the Daily Asterisk News? You have my permission. In a production environment any code will have gone through a testing phase, so your argument about a compiled language detecting stupid errors is pretty much irrelevant. Also most critical bugs will be in the script logic not the syntax. I'm also curious why you think the poor performance of a scripting language actually matters for AGI code? As stated most of the time will be spent by Asterisk streaming audio / waiting for a prompt. The few extra milliseconds a php script takes to start up are not going to be noticeable by any human listening to the call. Vince. -- _ -- 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] Connect mobile to asterisk
On Sat, 29 May 2010 11:31:05 +0530 (IST), Nivin Kumar nivinkuma...@yahoo.in wrote: I would like to connect my blackberry or any other cell phone to asterisk so that I can send calls through the sim card. I would also like to send SMS through this as well. Since wifi isn't as reliable and pervasive as GSM (and I read that BlackBerry don't allow VoIP clients anyway, so as to force users to make calls through their cellphone provider), I assume you don't want to connect the Blackberry to Asterisk through either through USB/Ethernet or wifi, but rather through GSM. The only solution I know is to buy a GSM gateway that will be connected by wire to your Asterisk server at home, and you'll need to get a second GSM subscription so that the GSM gateway has a SIM and let you connect your GSM phone to your Asterisk server. www.voip-info.org/wiki/view/VOIP+GSM+Gateways If someone knows of a cheaper way to connect a GSM phone to an Asterisk server, I'm also interested. -- _ -- 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
[asterisk-users] [Dahdi/system.conf] fxsks = 1 deprecated?
Hello I was editing files manually, and noticed that if I include the familiar fxsks=1 in /etc/dahdi/system.conf, Dahdi fails loading: = # cat /etc/dahdi/system.conf loadzone= fr defaultzone = fr fxsks=1 = # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] = Uncommenting that line solves the issue: = # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: [ OK ] = Does it mean it's no longer necessary to tell Dahdi which signaling to use for an FXO port? 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
Re: [asterisk-users] [Dahdi/system.conf] fxsks = 1 deprecated?
On Fri, 28 May 2010 10:45:34 -0500, Carlos Chavez cur...@telecomabmex.com wrote: Do you have a telephony card installed in that computer? Basically the error you get is because DAHDI cannot find any hardware that uses that signalling. When you comment it out it loads dahdi_dummy for timing. The wctdm module is for cards like the TDM400P from Digium or A400P from Openvox, if you do not have any of those cards in your computer then you need to load the proper module to activate the fxo port. Thanks for the tip. It's working now... except Dahdi was installed as asterisk.asterisk, while I need to install it as another user. Neither the linux/Makefile nor the README explain how to install Dahdi as another user. Does someone know? # ll /dev/dahdi/ total 0 crw-rw. 1 asterisk asterisk 196, 1 May 28 17:02 1 crw-rw. 1 asterisk asterisk 196, 2 May 28 17:02 2 crw-rw. 1 asterisk asterisk 196, 3 May 28 17:02 3 crw-rw. 1 asterisk asterisk 196, 4 May 28 17:02 4 crw-rw. 1 asterisk asterisk 196, 254 May 28 17:02 channel crw-rw. 1 asterisk asterisk 196, 0 May 28 17:02 ctl crw-rw. 1 asterisk asterisk 196, 255 May 28 17:02 pseudo crw-rw. 1 asterisk asterisk 196, 253 May 28 17:02 timer 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
Re: [asterisk-users] [Dahdi/system.conf] fxsks = 1 deprecated?
On Fri, 28 May 2010 17:57:24 +0200, Vincent codecompl...@free.fr wrote: Neither the linux/Makefile nor the README explain how to install Dahdi as another user. Does someone know? Found it: # vi /etc/udev/rules.d/dahdi.rules # /etc/init.d/dahdi restart # ll /dev/dahdi -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 12:29:05 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: This is a bug of the netjet module. It should not try to handle those devices. While they use the netjet chipset, they are not the ISDN BRI devices drivven by it. Thanks for the explanation. On this exact same hardware, I didn't have this problem with Dahdi/Zaptel 1.4. -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 11:41:09 +0200, Leonardo Pistone l.pist...@sispac.it wrote: Yes. dahdi_genconf reads /etc/dahdi/genconf_parameters and writes /etc/dahdi/system.conf and /etc/asterisk/dahdi_channels.conf. Thanks for the tip. Do you have asterisk installed? You neet at least to mkdir /etc/asterisk. Nope, and running mkdir /etc/asterisk solved this issue. There's one thing left: # /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting Running dahdi_cfg: [ OK ] I assume this reference to astribank is due to default settings. How can I remove unneeded drivers/modules? 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
Re: [asterisk-users] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 16:12:57 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Thanks for the explanation. On this exact same hardware, I didn't have this problem with Dahdi/Zaptel 1.4. Older kernel did not have the netjet module? Yup, that could be the reason. Anyway, problem solved :-) 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
Re: [asterisk-users] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 15:09:45 +0200, Vincent codecompl...@free.fr wrote: /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting Running dahdi_cfg: [ OK ] it's harmless. but it's a symtom of building dahdi-tools without libusb https://issues.asterisk.org/view.php?id=17189 -- _ -- 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
[asterisk-users] [X100P+Dahdi 2.3.0] Couple of questions
Hello, From www.x100p.com, I bought one of those cheap FXO cards. I have a couple of questions/issues about it: 1. I noticed that... - after cold booting the host, I see successful Dahdi/wcfxo messages in /var/log/messages - then, if I run either /etc/init.d/dahdi restart, or /etc/init.d/dahdi stop; /etc/init.d/dahdi start without waiting more than about 10 seconds between the stop/start commands, I get the familiar error messages DAHDI_CHANCONFIG failed on channel 1: No such device or address (6), Failed to initailize DAA, giving up error, and massive FXO PCI Master abort errors in /var/log/messages. According to this thread, this error with X100P cards can be due to some strange wiring: https://issues.asterisk.org/view.php?id=14232 http://www.mail-archive.com/asterisk-...@lists.digium.com/msg35317.html However, this occured on a host running Dahdi 2.3.0: Does it mean that this fix hasn't been ported from Zaptel to Dahdi, or that this error can have another cause? Could it be some timing issue in hardware and/or software, or maybe some initialization issue? In which case, is there a solution? IOW (and I don't mean this as criticism), is the X10xP hardware really crappy by design, or is the real cause for those problems to be found in the Zaptel code which were never really looked into because (understandably) developers prefered to work on the wctdm driver for the more professional TDM cards? 2. This card has the Silicon Labs Si3014/Si3034 chips which are supposed to support global line standards. I'm located in continental Europe, and apparently, for call-progress detection to have any chance to work correctly, I need to change the DAA from FCC (North America) to CTR21 (Europe). Does someone know how to do this? 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
[asterisk-users] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
Hello I'm trying to install Dahdi through source code on a Fedora 13 host to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv isn't happy. 1. After successfully running make all; make install; make config, I edited /etc/dahdi/system.conf thusly: loadzone=fr defaultzone=fr fxsks=1 2. Then ran dahdi_cfg -vv which says: - DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) - 3. So I ran lscpi -v: 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 20 I/O ports at a000 [size=256] Memory at e200 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel driver in use: netjet Kernel modules: wctdm, hisax, netjet FWIW, when I run modprobe wctdm followed by lsmod: # lsmod Module Size Used by wctdm 31892 0 dahdi 180789 1 wctdm netjet 12563 0 isdnhdlc3343 1 netjet crc_ccitt 1217 2 dahdi,isdnhdlc mISDNipac 28346 1 netjet mISDN_core 61414 3 netjet,mISDNipac I'm not sure whether I should use the wctdm driver or this netjet driver which I've never seen before. Could it be that dahdi_genconf modules added some ISDN-related items that I don't need? 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
Re: [asterisk-users] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Wed, 26 May 2010 17:17:08 +0200, Vincent codecompl...@free.fr wrote: I'm trying to install Dahdi through source code on a Fedora 13 host to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv isn't happy. More information, as I investigate: # vi /etc/modprobe.d/dahdi.blacklist.conf #blacklist wct4xxp #blacklist wcte12xp #blacklist wct1xxp #blacklist wcte11xp #blacklist wctdm24xxp #blacklist wcfxo blacklist wctdm #blacklist wctc4xxp #blacklist wcb4xxp # /etc/init.d/dahdi stop Unloading DAHDI hardware modules: done # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Wed, 26 May 2010 17:30:08 +0200, Vincent codecompl...@free.fr wrote: More information, as I investigate: For those having the same issue, here's what I learned: 1. In /etc/modprobe.d/dahdi.blacklist.conf, blacklist the netjet driver: blacklist netjet 2. To configure Dahdi, edit /etc/dahdi/system.conf: #For France loadzone= fr defaultzone = fr fxsks = 1 Next, start Dahdi... /etc/init.d/dahdi start ... and check /var/log/messages. DON'T RUN dahdi_genconf, as it overwrites system.conf. == I still have a couple of issues left: 1. When I run dahdi_genconf: /usr/sbin/dahdi_genconf: Failed to open /etc/asterisk/dahdi-channels.conf: No such file or directory 2. /etc/init.d/dahdi start: Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting Running dahdi_cfg: [ OK ] 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
[asterisk-users] Asterisk, NAT, and RTP?
Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct: - In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an SIP server, meaning it only acts as a registrar (for SIP end-points to make themselves know with an IP + RTP ports), and then as a Central office (to ring the other SIP end-point, and close the connection when an SIP end-point decides to hangup) - OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call transfer, call parking, etc.), it must remain in the loop, and hence, by default (canreinvite=no), all RTP packets always go through Asterisk, even if both SIP end-points live in the same network as the Asterisk server (and hence, since NAT is not involved, there's no need for any kung-fu with rewriting information in SDP packets and asking the NAT box to open the relevant ports for RTP) Is this correct? Thank you. -- _ -- 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] Starting and installing Dahdi (2.2.0)?
Hello Unless I overlooked it, the Asterisk Reference Information Version 1.6.1.6 at www.asterisk.org/docs doesn't include instruction on how to start Dahdi when used to drive a TDP PCI card (OpenVox A400P with a single FXO module www.openvox.cn/products/show.php?itemid=20lang=2). I'd like to know... 1. How to start Dahdi manually. Is this the right way? modprobe wctfxo modprobe wctdm modprobe zaptel 2. How to add a startup script in CentOS through chkconfig If this is covered in an up-to-date document on the Net, please tell me where it can be found. Thank you. ___ -- 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] Starting and installing Dahdi (2.2.0)?
It looks like make config takes care of installing an init script, so I can just run /etc/init.d/dahdi start to load the required modules. I get the following error, however: --- # /etc/init.d/dahdi start Loading DAHDI hardware modules: wcfxo: [ OK ] Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] --- FWIW, the PCI card seems to be correctly detected: --- # lspci -v 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 12 I/O ports at a000 [size=256] Memory at e200 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 --- Here's /etc/dahdi/system.conf: --- loadzone=fr defaultzone=fr fxsks=1 --- Any idea what is wrong? Thank you. ___ -- 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] Starting and installing Dahdi (2.2.0)?
... but ls -l /dev/dahdi/ doesn't return channel #1 :-/ # ls -l /dev/dahdi/ total 0 crw-rw 1 root root 196, 254 Dec 8 13:38 channel crw-rw 1 root root 196, 0 Dec 8 13:38 ctl crw-rw 1 root root 196, 255 Dec 8 13:38 pseudo crw-rw 1 root root 196, 253 Dec 8 13:38 timer # lsmod dahdi_dummy 8484 0 wcfxo 16032 0 dahdi 192392 2 dahdi_dummy,wcfxo # dahdi_cfg -vvv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) Could it be some incompatibily between this TDM card and the motherboard/PCI bus? ___ -- 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] Starting and installing Dahdi (2.2.0)?
I got it figured out: Modules must be listed in /etc/dahdi/modules: wcfxo wctdm dahdi /etc/init.d/dahdi start dahdi_cfg -vvv HTH, ___ -- 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] [DAHDI 2.2.0.2] failed on channel 1: No such device or address
Hello No matter if I use the entry-level OpenVox A400 with a single FXO module or the most-often-garbage card from ww.x100p.com, I get the following error after just adding this card to Mini-ITX with a single PCI slot (OS = CentOS 5.4): # lspci -v 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 12 I/O ports at a000 [size=256] Memory at e200 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 # cat /proc/interrupts CPU0 CPU1 0: 409671 0IO-APIC-edge timer 1: 2 0IO-APIC-edge i8042 7: 0 0IO-APIC-edge parport0 8: 3 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 7380 0IO-APIC-edge ide0 15: 3406 0IO-APIC-edge ide1 169:122 0 IO-APIC-level uhci_hcd:usb5, HDA Intel 201: 0 0 IO-APIC-level ehci_hcd:usb1, uhci_hcd:usb2 209: 0 0 IO-APIC-level uhci_hcd:usb3 217: 0 0 IO-APIC-level uhci_hcd:usb4 225:333 0 PCI-MSI eth0 NMI: 0 0 LOC: 412563 419484 ERR: 0 MIS: 0 # cat /etc/dahdi/system.conf loadzone=fr defaultzone=fr fxsks=1 # dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration --- Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) If someone's already seen this error message, any idea what the cause is, and what I could try to solve it? Thank you for any tip. ___ -- 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] [IAX] Recommended soft- and hardphones?
Hello Since SIP/RTP is a pain to use with road warriors who need to connect from any location over the Internet, I'd like to get them some IAX phones instead. For those of you using this protocol instead of SIP, what would you recommend as IAX hardphones and Windows (and ideally Mac) softphones? Thank you. ___ -- 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] Linux/Asterisk on game consoles?
On Fri, 16 Oct 2009 10:08:14 +0300, Stelios Koroneos skoron...@digital-opsis.com wrote: I did it with PS3 and Asterisk 1.2 about a year ago With Yellow Dog linux running on PS3 Was not using any of co-processors though, just the main cpu. Thanks for the feedback. ___ -- 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] Fax with a AEx410P and Beronet BN4S0 = Sending Problem
Hi all, I have problem with one of my configuration : FAX - AEX410P (One FXS port) --- BN4S0 PSTN Case 1 : Receiving Fax is Ok ( PSTN --- BN4SO -- AEX410P -- FAX ) Case 2 : Sending Fax is nok ( FAX --- AEX410P -- BN4SO -- PSTN ) I think we have some synchronisation problem because , de beginning of the fax is correct but I have some blank line on document and Asterisk do not release the line. This is the result of dahdi show channel 1 : Channel: 1LI File Descriptor: 13 Span: 1 Extension: Dialing: no Context: from-internal Caller ID: 70 Calling TON: 0 Caller ID name: device Mailbox: 7...@device Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook And this is configration of midn port : [PORT 2] - name: swisscom - allowed_bearers: all - far_alerting: no - rxgain: 0 - txgain: 0- te_choose_channel: no - pmp_l1_check: no - reject_cause: 21 - block_on_alarm: no - hdlc: no - context: ext-did-0002- language: en - musicclass: default - callerid: - method: standard - dialplan: 0 - localdialplan: 0 - cpndialplan: 0 - nationalprefix: 0- internationalprefix: 00 - presentation: -1 - screen: -1 - always_immediate: no - nodialtone: no - immediate: no- senddtmf: no - astdtmf: no - hold_allowed: no - early_bconnect: yes - incoming_early_audio: no - echocancel: 128 - need_more_infos: no - noautorespond_on_setup: no - nttimeout: no - bridging: yes- jitterbuffer: 4000 - jitterbuffer_upper_threshold: 0 - callgroup: - pickupgroup: - max_incoming: -1 - max_outgoing: -1 - l1watcher_timeout: 0 - overlapdial: 0 - msns: * - faxdetect: no- faxdetect_context: - faxdetect_timeout: 5 - ptp: no Somebody already have this problem ? Thanks for your precious help, Vincent ___ -- 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] Linux/Asterisk on game consoles?
Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you. ___ -- 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] Asterisk and Sheeva wall wart.
On Thu, 8 Oct 2009 08:40:37 -0400 (EDT), Ken D'Ambrosio k...@jots.org wrote: Hey, all. I'm seriously thinking about doing the VoIP thing at home. The perfect platform seemed to be the Sheeva wall wart I can't help you with the issue you had compiling for IAX, but I'm very interested in your experiment with getting Asterisk to run on the SheevaPlug. ___ -- 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] Connecting home intercom to Asterisk?
Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. Thank you. ___ -- 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] Connecting home intercom to Asterisk?
Yes - I have a similar access control using VoIP Pantel (Aleen) and Viking Units w- a C1000 module -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent Sent: Thursday, September 24, 2009 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Connecting home intercom to Asterisk? Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. Thank you. ___ -- 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] Asterisk on a Beagleboard?
Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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] GSM cellphone as cheap gateway?
Hello Even the cheapest GSM gateway I found for a simple SOHO server costs over $300. If possible, I'd rather have a local solution than use a remote VoIP provider to have Asterisk make outgoing calls to the GSM network. So... I was wondering if there are some entry-level cellphones that can somehow be hooked up to a PC running Asterisk and used as an el-cheapo GSM gateway? Thank you. ___ -- 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] Atcom AG188N as FXO?
Hello According to this article, this nice little unit can only use the PSTN port for outgoing calls (ie. as a backup in case the connection to the VoIP provider stops working), but not incoming calls: http://tinyurl.com/mwjmo8 Can someone confirm that Atcom made this strange decision, and that there's no work-around? Thank you. ___ -- 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] Web Browser Pop-up
Heelo, I currently search a program that can make a web browser Pop-up on an incoming call on a specific URL like : http://directorie.ch?CALLNUMBER:00451849799 I have found ADM, but it's a bit more complex for my purpose an it's not very stable. Do you know a simple software for that ? An other part of my project is to eneable click-to-call from a web page, do you know a kind of project that implement callto protocol, at this time I use Noojee click but It only work with Firefox. Thanks for your help, Vincent ___ -- 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] [extensions.conf] Any idea why not working as it should?
Hello
I noticed a small bug in the way my extensions.conf work:
Users can choose extensions 1-4 or 9 to tell why they're calling, and
I'll send an e-mail to the person(s) to whom is involved. Extension 4
is actually for personal messages for User1, and extension 9 is for
everyone (User1, User2, and User3).
= For some reason, when the caller chooses extension 4, both User1
and User2 get the e-mail, while I expect only User1 to get one:
= /usr/local/etc/asterisk/extensions.conf
//Caller can choose extensions 1, 2, 3, 4 or 9
//Look up software name from extension
exten = _[1-49],1,AGI(convert_app.phpcli|${EXTEN})
exten = _[1-49],n,Set(APPLICATION_NUM=${EXTEN})
//Send e-mail to who is in charge of the application
exten =
_[1-49],n,AGI(send_call_notification.phpcli|${CALLERID(name)}|${CALLERID(num)}|${APPLICATION_NUM})
exten = _[1-49],n,Hangup()
= /usr/local/share/asterisk/agi-bin/send_call_notification.phpcli
switch($argv[3]) {
//Softare 1
case 1:
$mail-AddAddress(User1);
break;
//Softwares 2,3
case 2:
case 3:
$mail-AddAddress(User2);
$mail-AddAddress(User1);
break;
//Personal msg
//BUG: Why does User2 also receive this e-mail?
case 4:
$mail-AddAddress(User1);
break;
//Any other subject
case 9:
$mail-AddAddress(User3);
$mail-AddAddress(User2);
$mail-AddAddress(User1);
break;
}
= Even when user selects extension 4, e-mail is sent to User1 and
User2!
To: us...@acme.com, us...@acme.com
Subject: [Personal msg for User1] Call from John Doe (555-1234)
===
Any idea why this is?
Thanks for any hint.
___
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Re: [asterisk-users] [Atcom] Asterisk + LAMP on 128MB RAM?
On Tue, 02 Jun 2009 02:17:02 +0800, Steve Underwood ste...@coppice.org wrote: Linux is not a given here. The Blackfin runs uCLinux, as it has non MMU. Don't get too enthusiastic about putting complex applications like Apache, MySQL or PHP on one of those boxes. The memory management limitations of uCLinux can be quite restricting. I'll keep that in mind, and see if works OK on that hardware. Thank you. ___ -- 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] [Atcom] Asterisk + LAMP on 128MB RAM?
Hello I'm thinking of selling an Asterisk server based on Atcom's IP02 solid-state unit with one FXO and one FXS ports: http://atcom.cn/En_products_IP02.htm By default, this unit based on a 400MHz Blackfin 532 chip only has 64MB RAM and 256MB of NAND flash. Those can be increased to 128MB and 1GB, respectively. Do you think I can install Linux + Asterisk + LAMP (replacing MySQL with Firebird, to avoid license costs) on the default specs, or will it be a bit short? Thank you. ___ -- 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] [Atcom] Asterisk + LAMP on 128MB RAM?
On Mon, 01 Jun 2009 10:40:56 +0100, Alan Lord (News) alansli...@gmail.com wrote: Check out the Astfin project (http://blog.astfin.org/?page_id=2). I'm guessing they have already done what you need... Thanks guys. The LAMP is only used to let the user see the call logs, so I just need PHP + DBMS (preferably Firebird, but SQLite can do too) for small use. ___ -- 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] [Atcom] Asterisk + LAMP on 128MB RAM?
On Mon, 1 Jun 2009 13:21:57 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: You may save yourself a lot of hassle just storing the CDRs in a plain text CSV file (which asterisk does for you), then parsing it with PHP directly. Thanks for the tip. I'll see if I can do without an SQL engine, although I'll probably need one. This is just for SOHO users, so big-time CDR isn't needed. Hopefully, a small Linux (uCLinux, AstFin, etc.) + LAMP + Asterisk will fit in 128MB RAM. ___ -- 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] Questions on X100P/X101P cards
On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Some X100P cards (e.g.: those that are based on SI3034, but not those basedon SI3035) support programmable impedance settings. Sadly the wcfxo driver does not support it. Fixing it should mostly be a matter of lifting some code from wctdm.c and adapting it. Shouldn't be much of an issue. Anybody wants to try that? (The cards I have at home are SI3035, sadly) A more interesting task would be to add support for some newer (soft/win-) modems. Anybody wants to try that? The wcfxo driver needs some love and care. Don't expect Digium to do that for you. They have more important stuff to do. Go and write your own device drivers. Thanks guys. So, provided the card has the right DAA chips to match the country in which it is used (FCC or CTR21), all it takes to use this hardware to handle a POTS line is patching Zaptel? IOW, the hardware itself is good enough for SOHO use? According to the following document, NovaVox (which no longer sells X100P cards) provides a Zaptel patch for cards sold by X100P.com to support non-FCC countries and UK CID: This document describes how to configure an Open Source IP PBX with an X100P Special Edition (SE) FXO PCI card installed to support Caller ID received from a UK BT PSTN line. The configuration requires implementing a patch for Asterisk®/Zaptel that was originally written for the UK but has also been known to work in other countries.[...] The Zaptel wcfxo driver has two user configurable modes of operation, FCC to support US line standards and CTR21 to support European line standards. The Silicon labs Si3012/Si3035 DAA chip used in the original Digium X100P card and low cost X100P clone cards only supports FCC mode. However, the Si3014/Si3034 DAA chip used on the X100P SE supports global line standards. X100P SE Setup Guide - Global Line Standards http://novavox.co.uk/support/x100p.html Richard Spencer supp...@novavox.co.uk ___ -- 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] Questions on X100P/X101P cards
On Thu, 07 May 2009 10:16:55 -0400, Jon Pounder j...@inline.net wrote: yeah I agree with the above - I never really found echo to ever be a problem, my only complaint was on some less than stellar cpu's I was having dtmf recognition problems. BTW, can someone explain to a libart major like me (;-)) where echo comes on in a telephone conversation? I seem to recall it's due to the length of the line between the CO and the local party, but I'm not sure. Is there a way to keep track of this issue, and overtime, to configure it to answer a call by expecting such and such echo, and thus, avoid starting sampling from scratch every time? Thank you. ___ -- 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] Questions on X100P/X101P cards
On Thu, 7 May 2009 13:40:20 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Another thing: their global-line-standard should basically (if properly written) resolve http://bugs.digium.com/view.php?id=11057 . Though I guess the new code will actually be in DAHDI, as Zaptel is frozen. Ah yes, I seem to remember Zaptel had to change their name to DAHDI for some reason. Is there a mailing list to ask for more information about Zaptel/DAHDI? http://lists.digium.com/mailman/listinfo/ ___ -- 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] Questions on X100P/X101P cards
Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the computer's CPU? 2. Are they all bad, no matter what chipset is used (Intel, Motoral, Ambient)? If not, which offer good enough quality to handle a single POTS line? 3. Why are they often bad quality? Because the driver itself is badly written? Because PC's don't have enough speed to handle the tasks using their own CPU (hard to believe, but I don't know)? Thank you. ___ -- 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] Questions on X100P/X101P cards
On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News) alansli...@gmail.com wrote: For a cheap backup to your VOIP service they do the job. I wouldn't use them for a proper system though. Thanks for the feedback. I have two more questions: 1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not doing much more than this and running Asterisk? 2. Is it good enough to handle a single FXO line for professional use? 3. Can you give me a pointer about which X100p you bought on eBay? AFAIK, there are three chipsets : Intel, Motorola, and Ambient. Using a $15 card over an $80 card is not insignificant because I could then sell a small server for $99. Thank you. ___ -- 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] Compact, fanless appliance?
On Wed, 6 May 2009 12:17:44 +0200, randulo spamsucks2...@gmail.com wrote: Those reading the thread amy be interested in Askozia pbx http://www.askozia.com/pbx/ Thanks for the link. ___ -- 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] Compact, fanless appliance?
On Mon, 4 May 2009 10:07:06 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Do you want to build your own? If so, you can put togther a 1GHz fanless VIA miniITX board, case (that will take a drive or flash IDE), memory and psu for well under £200. Same system has one PCI slot for a card that might take analogue, ISDN2 or ISDN30. Thanks for the tip. I'll see how much a complete VIA-based system costs. ___ -- 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] Compact, fanless appliance?
On Sun, 26 Apr 2009 12:51:01 +0100, Tim Panton t...@westhawk.co.uk wrote: I'm running asterisk 1.4 on an NSLU2 , only a couple of channels and minimal transcoding, but it seems fine and stable. £80 + usb storage Thanks guys for the tips on EdgePBX and the Linksys. Is the NSLU2 still sold, or has it been end-of-lifed? AFAIK, the modded router/NSA boxes to run Asterisk boild down to this: - Linksys NSLU2 - Linksys WRT54 - Planex MZK-W04NU Are there others I should know about? ___ -- 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] Compact, fanless appliance?
Hello For those SOHO customers (ie. at most, a couple of POTS/ISDN connections and simultaneous SIP calls) who'd rather not use a big, noisy PC to run Asterisk, I'd like to offer an alternative that has the following features: - not old hardware sold on eBay, ie. it must be up-to-date hardware sold by a company currently in business - compact, silent - has room for a 2.5 hard-disk, but if not, must provide a CompactFlash plug - ideally, room for a PCI card, possibly laid down with a riser to save space - total cost (shipping + VAT) 200 euro If it's cheaper and not much work, I don't mind buying the parts and putting the box together myself, but otherwise, I'd rather order a complete box, ready-to-use. What are my options to provide customers with that kind of solution? Thank you for any hint. ___ -- 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] Compact, fanless appliance?
Hello, On Thu, 23 Apr 2009 05:18:27 -0500 (CDT), Joe Greco jgr...@ns.sol.net wrote: Can you give us some clues as to why you have disqualified the fanless and/or embedded devices that are normally recommended on the list (Soekris, etc)? I haven't: I'd like to know what the options are. I'm looking for an up-to-date list of commercially-available compact solutions to run Asterisk, including those from Soekris, Atcom, etc. Thank you. ___ -- 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] Compact, fanless appliance?
On Thu, 23 Apr 2009 11:51:02 +0100, Steve Howes st...@geekinter.net wrote: http://tinyurl.com/df8qfm www.voip-info.org/wiki/view/Asterisk+embedded+systems Thanks Steve. I knew about this list, but I wanted to make sure there weren't other, more complete sources about the subject. So at this point, it seems like it boils down to this: Soekris PCEngines Atcom (IP01: £160+VAT) Herologic (HL-463: $259) uCpbx (235.00 EUR) AstBoxes (168.00 EUR) Gumstix HP Thinclient t5720 Thank you. ___ -- 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] Call Pickup Works w/Linksys ATA, not with Cisco 7940G
On Tue, 7 Apr 2009, George Pajari wrote: I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking from an ITSP (server has no PCI boards). *8 Call Pickup works fine from any of the phones connected using the Linksys SPA2102. *8 Call Pickup does not work from the Cisco 7940G phones (chan_sip.c:13977 handle_request_invite: Nothing to pick up for 000d6556-eeb3001c-76b88543-7f51d...@192.168.0.211) Seems someone else had the same problem back in 2004 and got no answer. http://lists.digium.com/pipermail/asterisk-users/2004-April/036869.html Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- 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] Asterisk is not designed for University with largeuser base?
On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is not designed for University with large user base?
Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- 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] VLC
Hi, I thought of this but I don't know where to intercept the file to convert it. The email is automatically sent, this is configured in voicemail.conf. I tried to change the mailcmd with a custom script thinking I would get the parameters passed to sendmail but it doesn't work. Has anybody an idea what the mail sending mechanism is? Regards vincent De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas Envoyé : mercredi, 11. mars 2009 14:35 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] VLC From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bex Vincent Sent: Wednesday, March 11, 2009 3:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VLC Hi All, When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: In voicemail.conf: format = ogg The result is as follow: [Mar 11 09:42:17] WARNING[24867]: format_ogg_vorbis.c:527 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams! -- x=0, open writing: /var/spool/asterisk/voicemail/default/0213149689/tmp/V2adNF format: ogg, 0x81ec648 -- User hung up [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:533 ogg_vorbis_tell: Telling is not supported on OGG/Vorbis streams! [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:514 ogg_vorbis_trunc: Truncation is not supported on OGG/Vorbis streams! -- Recording was 0 seconds long but needs to be at least 2 - abandoning Nothing gets recorded in the file. Has anybody done this before? Either get ogg work with voicemail or get VLC to read wav49. Cheers Vincent Why don't you just install SOX and convert the file that way? You could just do a command like this System(/usr/bin/sox in.ogg out.mp3) ___ -- 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] VLC
Hi Steve, I am not too much worried about the quality, but soon we will have 7000 users with voicemail. We don't have accurate statistics on the voicemail usage but it might well be that it will take a lot of space on the servers. Ogg has been implemented in asterisk but it seems that nobody uses it. Regards vincent -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Steve Edwards Envoyé : mercredi, 11. mars 2009 21:19 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] VLC On Wed, 11 Mar 2009, Bex Vincent wrote: When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: Would format = wav do? While the files are 10x bigger, it's still only 1mb per minute. Unless you have to retain the files for a very long time and/or have a huge number of users, I can't see spending the CPU time to compress and decompress something that will probably only be listened to once and discarded. Personally, the increase in fidelity is good enough reason to me. Try format = wav|wav49 and listen to the files with a decent set of speakers. I know the typical handset approaches two cans and a piece of wet string fidelity wise, but, since you say attached to an email, your users will hear the difference. 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
[asterisk-users] VLC
Hi All, When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: In voicemail.conf: format = ogg The result is as follow: [Mar 11 09:42:17] WARNING[24867]: format_ogg_vorbis.c:527 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams! -- x=0, open writing: /var/spool/asterisk/voicemail/default/0213149689/tmp/V2adNF format: ogg, 0x81ec648 -- User hung up [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:533 ogg_vorbis_tell: Telling is not supported on OGG/Vorbis streams! [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:514 ogg_vorbis_trunc: Truncation is not supported on OGG/Vorbis streams! -- Recording was 0 seconds long but needs to be at least 2 - abandoning Nothing gets recorded in the file. Has anybody done this before? Either get ogg work with voicemail or get VLC to read wav49. Cheers vincent ___ -- 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] oslec + dahdi
On Thu, 22 Jan 2009, troxlinux wrote: I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn I have installed oslec and loaded, but it doesn't work me with dahdi modinfo oslec filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko description:Open Source Line Echo Canceller Zaptel Wrapper author: David Rowe license:GPL srcversion: 13813ACD4A228F69FF4B5C1 depends: vermagic: 2.6.18-92.1.22.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4. oslec is a great great great software, with the version of zaptel 1.4.11 I had it installed and without anything of echo in my card TDM 400 I almost have the same enviroment as you, I basically run the following script to get oslec work with my tdm411 card. #!/bin/sh cd /usr/src wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2 tar xjf linux-2.6.28.tar.bz2 wget http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-2.1.0.2.tar.gz wget http://downloads.digium.com/pub/telephony/dahdi-linux/dahdi-linux-2.1.0.3.tar.gz tar zxvf dahdi-linux-2.1.0.3.tar.gz ln -s /usr/src/dahdi-linux-2.1.0.3 /usr/src/dahdi mkdir /usr/src/dahdi/drivers/staging cp -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging sed -i s|#obj-m += dahdi_echocan_oslec.o|obj-m += dahdi_echocan_oslec.o| /usr/src/dahdi/drivers/dahdi/Kbuild sed -i s|#obj-m += ../staging/echo/|obj-m += ../staging/echo/| /usr/src/dahdi/drivers/dahdi/Kbuild echo 'obj-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild cd /usr/src/dahdi make make install cd /usr/src tar zxvf dahdi-tools-2.1.0.2.tar.gz cd /usr/src/dahdi-tools-2.1.0.2 ./configure make make install Hope it helps. Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- 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] [Asus Eeebox] USB FXO adapter?
Hello I'm contemplating building an Asterisk voice server out of the compact Asus EeeBox: http://www.asus.com/products.aspx?l1=24l2=165 But they're so compact, they don't have a PCI slot to handle an analog phone line. I'd like to minimize footpring and cables: Besides analog/SIP boxes like Linksys (extra cables + transformer), does someone know of a USB adapter that is self-powered and could take an analog line as input, convert voice to SIP, and send packets through the USB port? Thank you. ___ -- 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] [Asus Eeebox] USB FXO adapter?
On Tue, 06 Jan 2009 16:51:40 +0100, Loic Didelot ldide...@mixvoip.com wrote: Use xorcom products: www.xorcom.com They provide usb devices for: fox, fxs, bri, pri Thanks but apparently, they don't have single-line USB devices, just a whole bank: www.xorcom.com/telephony-interfaces/telephony-interfaces.html ___ -- 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] [FreeBSD 6.3] Upgrading Zaptel messed up host
Hello Since the Ports collection showed that there were more recent versions of Asterisk and Zaptel, I tried to compile/install Zaptel, but it fails, even after stopping Zaptel cleanly, and even after stopping Asterisk itself, so I decided to just reboot. Now, when I type ztcfg -vv, I lose the SSH connection for a couple of minutes. === Dec 20 14:37:21 freebsd kernel: Zapata Telephony Interface Registered on major 196 Dec 20 14:37:21 freebsd kernel: Zaptel Version: zaptel-bsd-ng v0.0.1 Dec 20 14:37:21 freebsd kernel: Zaptel Echo Canceller: MG2 Dec 20 14:37:21 freebsd kernel: wctdm0 port 0xb400-0xb4ff mem 0xf500-0xf5000fff irq 9 at device 11.0 on pci2 Dec 20 14:37:21 freebsd kernel: wctdm0: [FAST] Dec 20 14:37:21 freebsd kernel: Freshmaker version: 71 Dec 20 14:37:21 freebsd kernel: Freshmaker passed register test Dec 20 14:37:21 freebsd kernel: Module 0: Installed -- AUTO FXO (FCC mode) Dec 20 14:37:21 freebsd kernel: Module 1: Not installed Dec 20 14:37:21 freebsd kernel: Module 2: Not installed Dec 20 14:37:21 freebsd kernel: Module 3: Not installed Dec 20 14:37:21 freebsd kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (1 modules) Dec 20 14:37:21 freebsd kernel: link_elf: symbol te11xp_init undefined === Any idea what's wrong? Thank you. ___ -- 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] Disable CDR?
Hello
I'm running Asterisk 1.4.21.2 on FreeBSD 6.3.
This part of extensions.conf...
;play a menu, and expect user to type any extension 1-4 or 9
exten = s,n,Wait(1)
exten = s,n,Background(main_menu)
exten = s,n,WaitExten(5)
exten = s,n,Hangup()
exten = _[1-49],1,AGI(convert_app.phpcli|${EXTEN})
... triggers this message:
-- Executing [EMAIL PROTECTED]:5] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:6] BackGround(Zap/1-1, main_menu) in
new stack
-- Zap/1-1 Playing 'main_menu' (language 'fr')
== CDR updated on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] AGI(Zap/1-1,
convert_app.phpcli|1) in new stack
I don't use CDR. Provided this will not have dire consequences, how
can I disable this?
Thank you.
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[asterisk-users] [1.4.21.2] Checking that already off-hook?
Hello Here's the scenario in my extensions.conf: 1. Check that CID is available 2. If not, go off-hook, and prompt the caller to type their CID number 3. Whether it was sent directly by the telco or input by the caller, look up the CID number if the DB, and rewrite the CID name on the fly 4. In the main menu, if not already off-hook, go off-hook; Then, play a menu to choose an extension So if the user calls with a CID number unmasked, once in Step 4, I need to check if the FXO card is already off-hook before playing the menu. What's a reliable way to check for this? Thank you. ___ -- 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] [1.4.21.2] Checking that already off-hook?
On Tue, 23 Sep 2008 12:23:28 +0200 (CEST), Julien Claassen [EMAIL PROTECTED] wrote: I wouldn't know a proper way to check for off-hook. But, couldn't you change your dialplan? Thanks for the suggestion, and this is how the script works now, but since most customers do call with CID enabled, I'd like to send a broadcast on the LAN to display this information on everyone's PC before Asterisk goes off-hook and does its spiel. Isn't there a way to check the status an FXO card is in? ___ -- 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] [1.4.21.2] Checking that already off-hook?
On Tue, 23 Sep 2008 12:29:22 +0200, Vincent
[EMAIL PROTECTED] wrote:
Isn't there a way to check the status an FXO card is in?
Apparently, it's OK to call Answer() even if the channel is already
open:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Answer
So I guess I can simplify things this way:
[my-ivr]
HELLO=false
exten = s,1,GotoIf($[${LEN(${CALLERID(num)})} = 0]?nocid,1:cid,1)
exten = nocid,1,Set(HELLO=true);
exten = nocid,n,Answer()
exten = nocid,n,Playback(my_sound_files/hello)
exten = nocid,n,Read(CALLERID(num),my_sound_files/no_cid,10)
exten = nocid,n,GotoIf($[${LEN(${CALLERID(num)})} 10]?cid,1)
exten = nocid,n,Hangup()
;If number in DB, rewrite CID name on the fly
exten =
cid,1,AGI(check_cid.phpcli|${CALLERID(num)}|${CALLERID(name)})
exten = cid,n,Goto(main_menu,s,1)
[main_menu]
;OK to call Answer() even if line already off-hook
exten = s,1,Answer()
exten = s,n,ExecIf($[${HELLO} = true],Playback,my_sound_files/hello)
exten = s,n,Background(my_sound_files/main_menu)
exten = s,n,WaitExten(5)
exten = s,n,Hangup()
Thank you.
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[asterisk-users] [CID] Unknown IE 18/21?
Hello Apparently, those are just warnings, but I'd like to know what those messages mean: [Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 18 [Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 21 Thank you. ___ -- 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] [FreeBSD 6.3/Ports] Make does nothing
On Sat, 13 Sep 2008 00:44:28 +0200, Vincent [EMAIL PROTECTED] wrote: I updated the Ports collection to compile the latest Asterisk, but after running make config, make just returns without doing anything: For those having the same problem: make clean ; make config ; make ; make deinstall ; make reinstall does the trick. I shouldn't have expected csup to remove stale stuff from previous compilings. ___ -- 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
[asterisk-users] [FreeBSD 6.3/Ports] Make does nothing
Hello I updated the Ports collection to compile the latest Asterisk, but after running make config, make just returns without doing anything: = # pkg_version -v | grep asterisk asterisk-1.4.20.1_1needs updating (port has 1.4.21.2_3) ^C # cd /usr/ports/net/asterisk # make # = There's nothing in /var/log/messages that would explain why this happens. FWIW, Asterisk is currently running on this host, but until I type make deinstall ; make reinstall, I guess it shouldn't be a problem. Any idea why this is happening? Thank you. ___ -- 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
[asterisk-users] [FreeBSD 6.3] Right-way to recover Zaptel?
Hello I'm running Asterisk 1.4.20.1 on a FreeBSD that I compiled from the Ports collection. It's the second time I'm having an issue with a FXO card and/or the Zaptel driver. I couldn't figure out what else to do, so I just rebooted the server, but I'd like to know what happened, and whether there's a less drastic solution. Here's some infos: === # /usr/local/etc/rc.d/zaptel stop zaptelkldunload: can't find file wcte12xp.ko: No such file or directory kldunload: can't find file wcte11xp.ko: No such file or directory kldunload: can't find file wct4xxp.ko: No such file or directory kldunload: can't find file wct1xxp.ko: No such file or directory kldunload: can't unload file: Device busy kldunload: can't find file wcfxo.ko: No such file or directory kldunload: can't find file tau32pci.ko: No such file or directory kldunload: can't find file qozap.ko: No such file or directory kldunload: can't unload file: Device busy Sep 6 19:11:12 freebsd kernel: kldunload: attempt to unload file that was loaded by the kernel # kldstat Id Refs AddressSize Name 19 0xc040 7a05b0 kernel 21 0xc0ba1000 5c304acpi.ko 121 0xc2d6c000 19000linux.ko 131 0xc3ba9000 32000zaptel.ko 171 0xc3c0d000 a000 wcfxs.ko # kldunload -i 13 kldunload: can't unload file: Device busy # kldunload -i 17 kldunload: can't unload file: Device busy === Thanks for any tip. ___ -- 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] [FreeBSD 6.3] Right-way to recover Zaptel?
On Sat, 06 Sep 2008 12:47:58 -0600, Anthony Francis [EMAIL PROTECTED] wrote: If Asterisk is running that will happen. Make sure to shutdown asterisk cleanly before doing that. Sorry, forgot to say that I couldn't restart or stop/start Asterisk: [Sep 6 19:06:17] WARNING[23110]: chan_zap.c:4157 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 [Sep 6 19:06:20] WARNING[23110]: chan_zap.c:4157 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 freebsd*CLI Disconnected from Asterisk server Executing last minute cleanups # asterisk -vr Asterisk 1.4.20.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. == Parsing '/usr/local/etc/asterisk/asterisk.conf': Found == Parsing '/usr/local/etc/asterisk/extconfig.conf': Found Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) # # reboot I couldn't unload the Zaptel driver manually, and couldn't restart Asterisk :-/ Is there something else I could have tried before rebooting? Thank you. ___ -- 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] Magnetic door locks
Yes, I used a Pap2 adaptor attached door tamperproof video/speaker phone. The model I used had alarm contacts just in case it was removed from the wall you can instant trigger an alarm system. You preprogram the extension it dials and it waits to here a touch tone code that NO NC contacts are activated to do with what you please. I do not remember specifically which door phone I used but here is a link to a exhibit which list several manufacturers. http://www.archiexpo.com/cat/home-building-automation-security/access-contro l-video-and-audio-door-phones-Y-652.html Vince APCN -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of c james Sent: Thursday, July 17, 2008 8:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Magnetic door locks I have an opportunity to interface asterisk with a security system to open their magnetic door locks. The security system needs a dry contact close upon activation to signal the door. Has anyone done this before? ___ -- 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] User unable to use DTMFs?
On Tue, 1 Jul 2008 04:23:19 -0700 (PDT), Benjamin Jacob [EMAIL PROTECTED] wrote: Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. Users call into our Asterisk voice server through a Zaptel PCI interface from regular phones, usually from a PBX (virtually all of them ISDN-based). The only files I modified are zaptel.conf, zapata.conf, and extensions.conf, which don't have anything DTMF-related, so Asterisk uses the default options. Thanks. ___ -- 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
[asterisk-users] [FreeBSD 6.3] Why not use safe_asterisk?
Hello I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case it crashes. Is this correct, and if yes, why not use it? Thank you. ___ -- 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] [FreeBSD 6.3] Zaptel stops responding
On Thu, 19 Jun 2008 11:36:27 +0200, Vincent [EMAIL PROTECTED] wrote: Will do, although it could be a problem in the Zaptel code, which is not written by the mfg. Thanks. I also notice that I can't restart the driver: # /usr/local/etc/rc.d/zaptel restart zaptelkldunload: can't unload file: Device busy zaptelkldload: can't load /usr/local/lib/zaptel/zaptel.ko: File exists ___ -- 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] [FreeBSD 6.3] Zaptel stops responding
On Wed, 18 Jun 2008 12:47:04 -0500, Tilghman Lesher [EMAIL PROTECTED] wrote: Please call the reseller from which you bought the card or the manufacturer for support. Will do, although it could be a problem in the Zaptel code, which is not written by the mfg. Thanks. ___ -- 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
[asterisk-users] [FreeBSD 6.3] Zaptel stops responding
Hello This PC had been running a Ports-compiled Asterisk 1.4.16.x succesfully for almost three months, but this morning, although Asterisk itself seemed fined, the Zaptel interface stopped taking calls. Stopping/restarting Zaptel using /usr/local/etc/rc.d/zaptel stop-start didn't let things recover. Since I didn't know better, I had to reboot the host to get things working. I used this opportunity to upgrade to Asterisk 1.4.20.1_1 and Zaptel 1.4.6_5. The hardware is an OpenVox PCI card with a single FXO module. I know, it's not a $200 Sangoma, but then, we only get a few calls/day. Has someone seen this before? Any idea what happened, and what to do to recuce the probabilty that it happens again? For instance, since it's a low-use host, (I don't mind running a CRON job to stop/start the driver a couple of times a day to keep things running. Thanks for any hint. ___ -- 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] Is Asterisk really good??
On Thu, 10 Apr 2008 11:46:48 -0700, Eugen Soare [EMAIL PROTECTED] wrote: So this is just a general question, Is Asterisk really good? Yes, but you should also look at an alternative that used Asterisk as a reference (www.freeswitch.org), and make an informed decision. ___ -- 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] [1.4.17] Disabled modules - Zaptel stops picking up calls
Hello I have a couple of questions about running 1.4.17 on FreeBSD 6.3: 1 .On a FreeBSD host, In modules.conf, I naively removed the following modules that I thought I didn't need, but after stopping/restarting Asterisk, Zaptel stops reporting calls: /usr/local/etc/asterisk/modules.conf noload = pbx_ael.so noload = res_smdi.so noload = chan_iax2.so = I don't use those features, so why would removing them affect Zaptel? 2. /usr/local/etc/rc.d/asterisk doesn't make use of the watchguard safe_asterisk, and just launches the asterisk binary directly. Why? Does someone have a modified copy of the script that uses safe_asterisk instead? Thank you. ___ -- 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] [1.4.17] Disabled modules - Zaptel stops picking up calls
On Thu, 10 Apr 2008 16:14:01 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote: chan_zap.so failed to load as it depends on res_smdi.so ? I have no idea. Is there an up-to-date list somewhere, or some script that lists dependencies for each module, so that we have some way of knowing what can be safely disabled? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hearing transfer during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word transfer, I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
[ht286]
type=friend
regexten=6010
username=ht286
secret=secret
context=numberplan-local
callerid=Home Phone 6010
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
[EMAIL PROTECTED]
dtmfmode=rfc2833
extensions.conf:
[macro-stdexten]
exten = s,1,Dial(${ARG2},20,t)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[default]
exten = s,1,Ringing
exten = s,n,Wait(1)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,Background(if-u-know-ext-dial)
exten = s,n,Background(otherwise)
exten = s,n,Background(to-reach-operator)
exten = s,n,Background(pls-hold-while-try)
exten = s,n,WaitExten(6)
exten = s,n,Hangup()
exten = i,1,Playback(invalid)
exten = i,n,Goto(s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup()
include = internal
[internal]
; define local extensions here
exten = 6010,1,Macro(stdexten,${EXTEN},SIP/ht286)
[numberplan-local]
ignorepat = 9
include = default
include = parkedcalls
comment = Local Calling
include = internal
features.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
;context = numberplan-local; Which context parked calls are in
;parkingtime = 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout = 3 ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
xfersound = beep; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot = next ; Continue to the 'next' parking
space. Defaults to 'first' available
;pickupexten = *8 ; Configure the pickup extension. Default is *8
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer = # ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record (a.k.a. Touch Monitor)
atxfer = * ; A
users.conf:
[6004]
fullname = Analog User 4
secret = 6004
email =
cid_number =
zapchan = 4
context = numberplan-local
hasvoicemail = yes
hasdirectory = yes
hassip = no
hasiax = no
hasmanager = no
callwaiting = no
threewaycalling = no
mailbox = 6004
hasagent = no
group = 2
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Re: [asterisk-users] SJphone behind NAT/Firewall without sound
On Thu, 3 Apr 2008 22:30:10 -0500, kazabe [EMAIL PROTECTED] wrote: I need connect some LAN stations with SJphone to an Asterisk Server published on Internet. [...] I dont manage the asterisk server. I just manage my proxy/firewall, and i need to my users can connect to that server. SIP works like FTP: One channel to manage calls, and a second one for data (audio): http://freshmeat.net/articles/view/2079/ Since Asterisk doesn't (yet) support STUN, to get audio packets to be received, you must configure the NAT firewall to let them in, and route them inside to the Asterisk server. This must match whatever is listed under /etc/asterisk/rtp.conf (you can reduce the range from 1-2 to eg. 1-10010; I could be wrong, but I think RTP actually needs two channels per call.) The same thing is required for the client hosts running the SJPhone application, but from what I read, most firewalls will work without having to map ports, and STUN-capable applications like SJPhone will keep the UDP ports open by sending out dummy packets regularly. If you can't modify the NAT firewall in front of the Asterisk server, I don't see how to solve this. ___ -- 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] Listening on conversations for training?
Hello I assume it's possible to do this with Asterisk: To train a new worker who works remotely, I'd like to have him listen in on support calls so that he gets to learn the kind of problems that come in, and how they're solved. When a call comes in and the support person thinks it's worthy to have the trainee be part of it, he will ring the trainee so he can join the call. From what I read, there seems to be two ways to do this: - either create a conference call, in which case the customer knows that a third party is part of the call - or have the trainee listen in on the conversation, unbeknownst to the customer Does someone use Asterisk for this purpose, and could tell me what the best solution is, and how to set things up? Thank you. ___ -- 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] Web page to show online extensions?
Hello Has someone written a web page (preferably PHP) that simply shows what extensions are currently online? Thank you. ___ -- 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] Web page to show online extensions?
On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger [EMAIL PROTECTED] wrote: http://www.micpc.com/eventmonitor/ Thanks guys. I was also thinking of stand-alone apps like Jabber or something. The call is simply to know if an extension is on- or offline. ___ -- 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] Hearing transfer during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word transfer, I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
[ht286]
type=friend
regexten=6010
username=ht286
secret=secret
context=numberplan-local
callerid=Home Phone 6010
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
[EMAIL PROTECTED]
dtmfmode=rfc2833
extensions.conf:
[macro-stdexten]
exten = s,1,Dial(${ARG2},20,t)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[default]
exten = s,1,Ringing
exten = s,n,Wait(1)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,Background(if-u-know-ext-dial)
exten = s,n,Background(otherwise)
exten = s,n,Background(to-reach-operator)
exten = s,n,Background(pls-hold-while-try)
exten = s,n,WaitExten(6)
exten = s,n,Hangup()
exten = i,1,Playback(invalid)
exten = i,n,Goto(s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup()
include = internal
[internal]
; define local extensions here
exten = 6010,1,Macro(stdexten,${EXTEN},SIP/ht286)
[numberplan-local]
ignorepat = 9
include = default
include = parkedcalls
comment = Local Calling
include = internal
features.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
;context = numberplan-local; Which context parked calls are in
;parkingtime = 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout = 3 ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
xfersound = beep; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot = next ; Continue to the 'next' parking
space. Defaults to 'first' available
;pickupexten = *8 ; Configure the pickup extension. Default is *8
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer = # ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record (a.k.a. Touch Monitor)
atxfer = * ; A
users.conf:
[6004]
fullname = Analog User 4
secret = 6004
email =
cid_number =
zapchan = 4
context = numberplan-local
hasvoicemail = yes
hasdirectory = yes
hassip = no
hasiax = no
hasmanager = no
callwaiting = no
threewaycalling = no
mailbox = 6004
hasagent = no
group = 2
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[asterisk-users] Calling extension from CLI?
Hello For testing purposes, is it possible to call an extension from the command-line interface, just so I can trigger calls to AGI scripts from a test extension? Thank you. ___ -- 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] Access rights between AGI and Web server?
On Sun, 23 Mar 2008 19:55:32 -0600, Chris Carey [EMAIL PROTECTED] wrote: Correction: I run the web server and asterisk both as the user asterisk I wish I could, but I have no idea how to safely tell Asterisk to run as www instead of root, as it does now. I assume I'll have to chmod/chown a bunch of files and directories, but I'd have to know exactly what to do. ___ -- 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] Access rights between AGI and Web server?
On Mon, 24 Mar 2008 11:05:32 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
If the AGIs do run as root:wheel, then there should be no problem,
because they should be able to access the db files?
I agree, but even after uninstalling Lighttpd and installing Apache2,
just to make sure it weren't some security issue that would prevent a
PHP script from writing to files outside the /data directory, I have
the same issue :-/
?php
$u = posix_getpwuid(posix_getuid());
$g = posix_getgrgid(posix_getgid());
echo This script is running as .$u['name'].:.$g['name'];
?
1. Here's the output:
echo exec('id') . hr;
$u = posix_getpwuid(posix_getuid());
$g = posix_getgrgid(posix_getgid());
echo This script is running as .$u['name'].:.$g['name'];
=
uid=80(www) gid=80(www) groups=80(www)
This script is running as www:www
2. The PHP script and the SQLite database are owned by www:www:
[/usr/local/www/apache22/data]# ll
drwxr-xr-x 2 root wheel 512 Mar 24 19:52 .
drwxr-xr-x 6 root wheel 512 Mar 24 18:56 ..
-rw-r--r-- 1 www www2463 Mar 24 20:00 test.php
[/usr/local/share/asterisk/agi-bin]# ll
drwxr-xr-x 3 root wheel512 Mar 24 18:38 .
drwxr-xr-x 9 root wheel512 Mar 14 08:05 ..
-rw-rw-r-- 1 www www 3072 Mar 24 18:37 test.sqlite
3. And here's the code:
//GOOD $dbh = new PDO(sqlite:test.sqlite);
//GOOD $dbh = new PDO(sqlite:/tmp/test.sqlite);
$dbh = new
PDO(sqlite:/usr/local/share/asterisk/agi-bin/test.sqlite);
$time = time();
$current = date(Y-m-d H:i:s,$time);
$sql = INSERT INTO mytable VALUES (NULL,'$current');
print $sqlhr;
$dbh-exec($sql);
$sql = SELECT * FROM mytable;
foreach($dbh-query($sql) as $row) {
print $row['name'] . p\n;
}
$dbh = null;
I don't understand why test.php can read, but cannot write.
Thank you.
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Re: [asterisk-users] Access rights between AGI and Web server?
On Mon, 24 Mar 2008 12:09:00 -0800, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Now, that was run under a webserver. right? not under asterisk as an AGI? I thought we were expecting to see root:wheel :) Yup, sorry about: I forgot to say that I use a single SQLite database to share data between Asterisk and some PHP scripts. Found what it was: Even if a file is set to 664 and owned by the right user, the _directory_ in which the file lives has precedence. In this case, I just chowned it to root:www, and chmoded it to 664: [/usr/local/share/asterisk/agi-bin]# ll drwxrwxr-x 3 root www 512 Mar 24 22:05 . drwxr-xr-x 9 root wheel512 Mar 14 08:05 .. -rw-rw-r-- 1 www www 3072 Mar 24 22:05 test.sqlite Learned something new today. Thanks for the help. ___ -- 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] Access rights between AGI and Web server?
Hello I run AGI scripts from extensions.conf to save data into an SQLite database file, but this file must also be accessible in read-write mode by PHP scripts served by Lighttpd. As far as I can tell, Asterisk runs by default as root:wheel. I don't know if AGI scripts also run as root:wheel. Lighttpd runs as www:www, and if I create a new SQLite database through PHP scripts, they're created as www:wheel. What do you recommend I do so both AGI scripts and PHP scripts can work with a common SQLite file? Should I run Asterisk as www:www, www:wheel? Something else? Thank you. ___ -- 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] How Do I continue after Dial Command ??
On Sun, 9 Mar 2008 03:11:58 +0100, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: Two ways, use n priority or add 'g' iption in dial command. 2008/3/9, Jim Duda [EMAIL PROTECTED]: How do I get a context to continue to execute commands after the caller hangs up after a Dial command? I'm using the e option to the Dial application. I though the e option would allow the context to continue. This doesn't want to work for me. I'm using asterisk-1.6.beta5 I never get to 3 below. I get a message saying the 2 ended with a non-zero status. [sphinx] exten = s,1,AGI(MisterHouse.agi,Sphinx Connect) exten = s,2,Dial(CONSOLE/1,,e) exten = s,3,AGI(MisterHouse.agi,Sphinx Disconnect) exten = s,4,Hangup What about simply adding the h extension? [sphinx] exten = s,1,AGI(MisterHouse.agi,Sphinx Connect) exten = s,n,Dial(CONSOLE/1,,e) exten = s,n,AGI(MisterHouse.agi,Sphinx Disconnect) exten = s,n,Hangup exten = h,1,Verbose(Here we are) exten = h,n,Verbose(Next step) ___ -- 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] How Do I continue after Dial Command ??
On Sun, 09 Mar 2008 17:21:47 -0400, Jim Duda [EMAIL PROTECTED] wrote: exten = s,1,AGI(MisterHouse.agi,Sphinx Connect) exten = s,2,Dial(CONSOLE/1) Unless there's a technical reason for this, you should use n, so you can easily add/remove instructions without having to renumber everything: From exten = h,1,AGI(MisterHouse.agi,Sphinx Disconnect) exten = h,2,Hangup To exten = h,1,AGI(MisterHouse.agi,Sphinx Disconnect) exten = h,n,Hangup ___ -- 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] Newbie on VoIP
On Mon, 3 Mar 2008 10:14:02 +0800, NOC Ph [EMAIL PROTECTED] wrote: This questions might annoyed experts. Please bear with me... The journey of a thousand miles begins with a single step. Lao Tzu. Free PDF of Asterisk: The Future of Telephony, Second Edition http://downloads.oreilly.com/books/9780596510480.pdf ___ -- 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] How to get a clean, basic configuration?
On Tue, 26 Feb 2008 01:19:23 +0200, Atis Lezdins [EMAIL PROTECTED] wrote: To help you on your way of minimizing modules, here's some basic setup that generally works Thanks much for sharing your modules.conf. ___ -- 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] How to get a clean, basic configuration?
On Fri, 22 Feb 2008 09:15:35 -0600, Tilghman Lesher [EMAIL PROTECTED] wrote: Generally, the rule is that you can't remove any of the res_* modules. Thanks for the tip. At this point, I have the following in modules.conf, but when I type reload, it still loads stuff I disabled such as DunDI: == [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so load = res_musiconhold.so noload = chan_alsa.so noload = pbx_ael.so noload = pbx_dundi.so noload = res_config_pgsql.so noload = res_smdi.so [Feb 25 16:56:09] NOTICE[6763]: pbx_ael.c:4114 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. -- Reloading module 'pbx_dundi.so' (Distributed Universal Number Discovery (DUNDi)) == Parsing '/etc/asterisk/dundi.conf': Found == Moving the noload lines before autoload makes no difference. ___ -- 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] How to get a clean, basic configuration?
On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen [EMAIL PROTECTED] wrote: For the brave: use modules.conf without 'autoload = yes'. This promises you many hours of interesting dialplan debugging. Enjoy. Yup, that's what I anticipated, which is why I was asking which modules I can _safely_ remove without breaking things :-) ___ -- 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] How to get a clean, basic configuration?
On Thu, 21 Feb 2008 08:33:20 -0500, C F [EMAIL PROTECTED] wrote: first off I anwered you to use vi and you complained showing me cat. There's some misunderstanding. I didn't complain. I just didn't know if Asterisk only looked for stuff in modules.conf because there was so little there and so much stuff scrolling by when I type reload. Before loading modules explicitely, I need to make sure what each does precisely, and what the interdependencies are, if any, so that I know what the consequences are if I decide not to load a mdoule that looks like it's not needed on my setup. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite question
On Fri, 22 Feb 2008 18:50:16 +0800, Ron [EMAIL PROTECTED] wrote: If i set, canreinvite=yes on all ext, assuming all ip phones have the same codec, if 100 calls 101, or vice versa will rtp still go thru asterisk? and same scenario for 200 to 202 or vice versa. ... and I'd like to add to this question: If the phones have the option Enable NAT, I expected them to be able to talk to each other directly, but they didn't, and I had to set them to canreinvite=no in sip.conf. Why is that? ___ -- 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] How to get a clean, basic configuration?
On Thu, 21 Feb 2008 15:00:15 +1100, Paul Hales [EMAIL PROTECTED] wrote: Head off into /etc/asterisk/modules.conf and add some 'noload' lines. Ah, makes sense. Asterisk loads everything, and must be told explicitely _not_ to load something :-) Is there a comprehensive list that explains what each and every module does, so that I know what I can safely not load? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get a clean, basic configuration?
Hello I'm using a standard Asterisk install with default settings, and when I run reload, I see that Asterisk fetches configuration information from a lot more sources than just my extensions.conf and sip.conf. For instance: -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to 'us' == Parsing '/etc/asterisk/features.conf': Found == Parsing '/etc/asterisk/adsi.conf': Found == Parsing '/etc/asterisk/dundi.conf': Found == Parsing '/etc/asterisk/extensions.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4094 pbx_load_module: Starting AEL load process. [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4101 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. etc. How can I go and trim things down? Thank you. ___ -- 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] How to get a clean, basic configuration?
On Wed, 20 Feb 2008 21:44:30 -0500, C F [EMAIL PROTECTED] wrote: vi /etc/asterisk/modules.conf Thanks, but this file doesn't hold much that's uncommented by default: # cat /etc/asterisk/modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so load = res_musiconhold.so noload = chan_alsa.so Is this really the only file that Asterisk reads to know what to load? ___ -- 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] [Linux/Python 2.4.2] Forking Python doesn't work
On Wed, 13 Feb 2008 22:26:16 -0500, Russell Bryant [EMAIL PROTECTED] wrote: The arguments to System() are a bit different. Put it in just like you would type at the command line. System(/tmp/netcid.py 2000 Joe) That did it :-) Thanks guys. BTW, for those interested, I didn't have to double-fork: == #!/usr/bin/python import socket,sys,time,os sys.stdout = open(os.devnull, 'w') if os.fork(): sys.exit(0) else: #Here, send broadcast == ___ -- 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
