Re: [asterisk-users] SIP Hardphone that works well with asterisk
On Sat, 12 Feb 2011, ast guy wrote: Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Of-course, there's always good old Grandstream... Read the archives for lots of for's and againsts... I'm one of the 'for's' ... Just deployed a small number of their new GXP2100's too. I think this is their replacement for the GXP2000. They seem to be going for the Sit up and beg type phone rather than the flat desktop of the GXP2000. They're rather nice - a bit heavier than the GXP2000 and sound quality is better too. Also cheaper with the same (and more) functionality. I've not yet gotten my head round all their fancy XML screen setup stuff, but the built in weather app. does seem to work when you put in he right town code (from yahoo, apparently!) But a lot of people got stung with Grandstream in their early days, so it's buyer beware here. Personally I've gotten on really well with them and they've been a boon for clients who've been on a really tight budgets (I have a few charities, etc. on my books) I've also just re-cycled an office worth of 20 of them which are now 3.5 years old, and they looked a bit worn - the numbers seem to wear out on the buttons, but they were all functional. (after a cleanup!) Watch out for the PSUs though - I have had a higher than expected number of failures there, but I suspect it was down to one batch of dodgyness. Gordon -- _ -- 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] [modules.conf] Modules still loaded after noload
Hello I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I don't need: cat modules.conf noload = codec_speex.c ip04*CLI reload ip04*CLI show modules codec_speex.so Just to check, I added the actual filename (.so): cat modules.conf noload = codec_speex.c noload = codec_speex.so ip04*CLI reload ip04*CLI show modules codec_speex.so /etc/init.d/asterisk stop /etc/init.d/asterisk start asterisk -vvvr ip04*CLI show modules codec_speex.so Does someone know why Asterisk still loads modules even with the above lines in modules.conf? 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] Fax for Asterisk SIP-TDM
On 02/12/2011 10:53 PM, Mark Willis wrote: Is it possible to do SIP-Asterisk-TDM in a single step with FFA? Or does FFA always use TIFF files? I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102 ATA's at the fax machines and send faxes directly over a PRI. Asterisk does not currently support T.38-TDM gateway mode for FAX, although there is a patch on the issue tracker to add support for it, and it's in the works for Asterisk 1.10. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] [modules.conf] Modules still loaded after noload
On 11-02-13 09:52 AM, Gilles wrote: I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I don't need: Does someone know why Asterisk still loads modules even with the above lines in modules.conf? It looks like you're loading Asterisk, which loads all the modules, then modifying modules.conf and just doing a reload at that point. Try either restarting Asterisk to see if the modules still load (it shouldn't). Before doing the reload, I'd do a module unload chan_speex.so then do your reload and see if that works. I'm not sure reload actually looks at modules.conf at that point. It probably just reloads all the modules you have in memory, rather than unloading everything, then parsing modules.conf and loading everything in there back into memory (which I think is what you're expecting). Leif. -- _ -- 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] [modules.conf] Modules still loaded after noload
On 02/13/2011 09:36 AM, Leif Madsen wrote: I'm not sure reload actually looks at modules.conf at that point. It probably just reloads all the modules you have in memory, rather than unloading everything, then parsing modules.conf and loading everything in there back into memory (which I think is what you're expecting). This is correct. 'reload' is not 'restart', it only tells all the currently-loaded modules to 'reload' themselves (which generally means they will reparse their configuration files to look for changes). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] SIP Hardphone that works well with asterisk
My take on this is to not skimp on the phones. This is how people relate to the phone system you install. Good phones will, to them, imply a good system. And vise-versa. -- _ -- 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] SIP Hardphone that works well with asterisk
On Sun, 13 Feb 2011 09:28:19 -0700, Joel Maslak wrote: My take on this is to not skimp on the phones. This is how people relate to the phone system you install. Good phones will, to them, imply a good system. And vise-versa. Life is just too sort to suffer through using a cheap phone. The memory of the bargain fades long before the frustration of using the device, or the sense that it's just cheap. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] Call Files, Variable passing
Try Set instead of SetVar. On Sat, Feb 12, 2011 at 9:59 PM, Dan Dan dani.mani...@gmail.com wrote: Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, Action: login\r\n); fputs($oSocket, Events: off\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Exten: 3001\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Priority: $strPriority\r\n); fputs($oSocket, MaxRetries: $strMaxReTry\r\n); fputs($oSocket, RetryTime: $strRetryTime\r\n); fputs($oSocket, SetVar: DIAL1=$number1\r\n); fputs($oSocket, SetVar: DIAL2=$number2\r\n); fputs($oSocket, SetVar: AcceptParallel=$ap\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); Here I am trying to set three variables but they do not seem to be passed on to the extensions for dialing Am I following the right syntax ? Thanks -dani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk SIP-TDM
Dear I had good experience with asterisk + spandsp for sending and receiving fax, if your ip phone supports fax, you need asterisk only as g711(no vad) gateway. best On Sun, Feb 13, 2011 at 7:00 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/12/2011 10:53 PM, Mark Willis wrote: Is it possible to do SIP-Asterisk-TDM in a single step with FFA? Or does FFA always use TIFF files? I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102 ATA's at the fax machines and send faxes directly over a PRI. Asterisk does not currently support T.38-TDM gateway mode for FAX, although there is a patch on the issue tracker to add support for it, and it's in the works for Asterisk 1.10. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [modules.conf] Modules still loaded after noload
On Sun, 13 Feb 2011 10:36:43 -0500, Leif Madsen leif.mad...@asteriskdocs.org wrote: Try either restarting Asterisk to see if the modules still load (it shouldn't). Before doing the reload, I'd do a module unload chan_speex.so then do your reload and see if that works. Thanks for the tip, but I also tried restarting Asterisk, and codec_speex.so is still loaded: /etc/init.d/asterisk stop /etc/init.d/asterisk start Here's my modules.conf: ... noload = codec_speex.c Could there be some configuration somewhere that tells Asterisk to ignore modules.conf? 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] [modules.conf] Modules still loaded after noload
On 2011-02-13 15:21, Gilles wrote: noload = codec_speex.c Try noload = codec_speex.so Mark -- Mark Willis Star One Telecom Office: 1-800-889-7001 Cell: 210 880 5097 http://staronetel.com -- _ -- 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] [modules.conf] Modules still loaded after noload
On Sun, 13 Feb 2011 15:32:03 -0600, Mark Willis marksli...@markwillis.net wrote: Try noload = codec_speex.so That dit it. However, I'm puzzled by the fact that the default filenames in modules.conf all ended with .c instead of .so: === /etc/asterisk cat modules.conf [modules] autoload=yes noload = pbx_gtkconsole.c noload = pbx_kdeconsole.c noload = app_intercom.c ... === 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] [modules.conf] Modules still loaded after noload
On 2011-02-13 15:49, Gilles wrote: On Sun, 13 Feb 2011 15:32:03 -0600, Mark Willis marksli...@markwillis.net wrote: Try noload = codec_speex.so That dit it. However, I'm puzzled by the fact that the default filenames in modules.conf all ended with .c instead of .so: === /etc/asterisk cat modules.conf [modules] autoload=yes noload = pbx_gtkconsole.c noload = pbx_kdeconsole.c noload = app_intercom.c ... === Thank you. The are .so in every modules.conf I've seen. Mark -- Mark Willis Star One Telecom Office: 1-800-889-7001 Cell: 210 880 5097 http://staronetel.com -- _ -- 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] Question about EuroBRI final 2 digits
On 2/10/11 5:54 AM, Christian Gansberger christian.gansber...@accm.at wrote: Hello, Maybe try that: In your incoming isdn context: [isdn-incoming] exten = s,1,Set(TIMEOUT(digits)=3) exten = s,2,WaitExten(2) exten = s,3,Dial(SIP/operator...) exten = 10,1,Dial(SIP/10) exten = 20,1,Dial(SIP/20) So if a call comes in Asterisk waits, 2 seconds for further digits dialed and if so jumps to the right extension in the context. Overlapdial should be yes. yours christian gansberger www.accm.at Many thanks for this idea, Christian I have put this equivalent into the dialplan And when the Austria team gets to the office in the morning they will test it. (BTW changed TIMEOUT(digits) to TIMEOUT(digit)). Cassius On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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] [modules.conf] Modules still loaded after noload
On Sun, 13 Feb 2011 15:59:52 -0600, Mark Willis marksli...@markwillis.net wrote: The are .so in every modules.conf I've seen. You're right. It looks like the modules.conf that came with the Asterisk package is totally wrong: www.voip-info.org/wiki/view/Asterisk+config+modules.conf 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] Using files .call or AMI
Thks, now I understand for your cooperation.TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sat, 12 Feb 2011 23:20:11 + From: ro...@firedrake.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? I'm using this to place calls from local IP-phones over the PSTN. So my script will generate, say: Channel: SIP/lanphone Context: from-lan Extension: 08001234567 taking the 0800... from the list of customer details. SIP/lanphone is the ID of the originating phone. Extension is the sequence the agent would dial if he were placing the call himself. The originating phone rings; when it's picked up, the Asterisk server calls the Extension number and bridges the two calls, so the local agent hears ringing tones from the far end. All the agent has to do is pick up the phone when it rings and put it down when the call is over. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR with unix time.
On Thursday 10 February 2011 12:33:40 Rodrigo Lang wrote: 2011/2/10 Tilghman Lesher tilgh...@meg.abyt.es On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote: I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? The answer is, it depends upon the backend version you're using. With cdr_pgsql and cdr_mysql from 1.6.2 forward, if the column type is integer or float, then the unix timestamp will be used. Without any modification? Only with the column type, Asterisk will modify the common date to unix time? The idea behind this is that we don't want to lose any information. Thus, if the datatype is numeric, then the only way to ensure that we don't lose information during the insert is to set the data to a unixtime format. Note that we can even store fractions of a second in this way, if the column type supports it (i.e. decimal or float). -- Tilghman -- _ -- 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] sangoma wanpipe install error
Here's the messages log. There's a line that says ERROR: Unsupported DS E1 CHIP (00:00) Feb 14 10:09:19 server14 kernel: [6011515.237242] dahdi: Telephony Interface Registered on major 196 Feb 14 10:09:19 server14 kernel: [6011515.237246] dahdi: Version: 2.4.0 Feb 14 10:09:19 server14 kernel: [6011515.244707] WANPIPE(tm) Hardware Support Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:09:19 server14 kernel: [6011515.245010] usbcore: registered new interface driver sdlausb Feb 14 10:09:19 server14 kernel: [6011515.251320] WANPIPE(tm) Interface Support Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:09:19 server14 kernel: [6011515.262373] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:09:19 server14 kernel: [6011515.262376] wanpipe: Probing for WANPIPE hardware. Feb 14 10:09:19 server14 kernel: [6011515.264214] wanpipe: AFT-A104-SH PCIe T1/E1 card found (HDLC (DS) rev.37), cpu(s) 1, line(s) 4, bus #4, slot #4, irq #16 Feb 14 10:09:19 server14 kernel: [6011515.264228] wanpipe: Allocating maximum 4 devices: wanpipe1 - wanpipe4. Feb 14 10:09:19 server14 kernel: [6011515.264775] WANPIPE: TDM Codecs Initialized Feb 14 10:09:19 server14 kernel: [6011515.271478] WANPIPE(tm) Socket API Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:09:19 server14 kernel: [6011515.271481] NET: Registered protocol family 25 Feb 14 10:09:19 server14 kernel: [6011515.283341] WANPIPE(tm) WANEC Layer 3.5.18.0 (c) 1995-2006 Sangoma Technologies Inc. Feb 14 10:09:19 server14 kernel: [6011515.283344] wanec_create_dev: Registering Wanpipe ECDEV Device! Feb 14 10:10:32 server14 kernel: [6011588.876265] wanec_remove_dev: Unregistering Wanpipe ECDEV Device! Feb 14 10:10:32 server14 kernel: [6011588.876346] WANEC Layer: Unloaded Feb 14 10:10:32 server14 kernel: [6011588.879218] af_wanpipe: Unregistering Wanpipe API Socket Module Feb 14 10:10:32 server14 kernel: [6011588.904829] NET: Unregistered protocol family 25 Feb 14 10:10:32 server14 kernel: [6011588.907606] WANPIPE: TDM Codecs unloaded. Feb 14 10:10:32 server14 kernel: [6011588.907609] Feb 14 10:10:32 server14 kernel: [6011588.907610] wanpipe: WANPIPE Modules Unloaded. Feb 14 10:10:32 server14 kernel: [6011588.912729] usbcore: deregistering interface driver sdlausb Feb 14 10:10:32 server14 kernel: [6011588.912756] Logger TASKQ Not Running Feb 14 10:10:32 server14 kernel: [6011588.914895] dahdi: Telephony Interface Unloaded Feb 14 10:11:47 server14 kernel: [6011663.492120] dahdi: Telephony Interface Registered on major 196 Feb 14 10:11:47 server14 kernel: [6011663.492123] dahdi: Version: 2.4.0 Feb 14 10:11:47 server14 kernel: [6011663.500161] WANPIPE(tm) Hardware Support Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:11:47 server14 kernel: [6011663.500557] usbcore: registered new interface driver sdlausb Feb 14 10:11:47 server14 kernel: [6011663.505314] WANPIPE(tm) Interface Support Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:11:47 server14 kernel: [6011663.513738] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:11:47 server14 kernel: [6011663.513741] wanpipe: Probing for WANPIPE hardware. Feb 14 10:11:47 server14 kernel: [6011663.514985] wanpipe: AFT-A104-SH PCIe T1/E1 card found (HDLC (DS) rev.37), cpu(s) 1, line(s) 4, bus #4, slot #4, irq #16 Feb 14 10:11:47 server14 kernel: [6011663.514996] wanpipe: Allocating maximum 4 devices: wanpipe1 - wanpipe4. Feb 14 10:11:47 server14 kernel: [6011663.515447] WANPIPE: TDM Codecs Initialized Feb 14 10:11:47 server14 kernel: [6011663.522007] WANPIPE(tm) Socket API Module 3.5.18.0 (c) 1994-2010 Sangoma Technologies Inc Feb 14 10:11:47 server14 kernel: [6011663.522011] NET: Registered protocol family 25 Feb 14 10:11:47 server14 kernel: [6011663.534362] WANPIPE(tm) WANEC Layer 3.5.18.0 (c) 1995-2006 Sangoma Technologies Inc. Feb 14 10:11:47 server14 kernel: [6011663.534366] wanec_create_dev: Registering Wanpipe ECDEV Device! Feb 14 10:11:47 server14 kernel: [6011663.543452] wanpipe1: Starting WAN Setup Feb 14 10:11:47 server14 kernel: [6011663.543455] Feb 14 10:11:47 server14 kernel: [6011663.543457] Processing WAN device wanpipe1... Feb 14 10:11:47 server14 kernel: [6011663.543461] wanpipe1: Locating: A101/1D/A102/2D/4/4D/8 card, CPU A, PciBus=4, PciSlot=4 Feb 14 10:11:47 server14 kernel: [6011663.543467] wanpipe1: Found: A101/1D/A102/2D/4/4D/8 card, CPU A, PciBus=4, PciSlot=4, Port=0 Feb 14 10:11:47 server14 kernel: [6011663.543718] wanpipe1: AFT PCI memory at 0xFAFE Feb 14 10:11:47 server14 kernel: [6011663.543721] wanpipe1: IRQ 16 allocated to the AFT PCI card Feb 14 10:11:47 server14 kernel: [6011663.543733] wanpipe1: Starting AFT 2/4/8 Hardware Init. Feb 14 10:11:47 server14 kernel: [6011663.543741] wanpipe1: Enabling front end link monitor Feb 14 10:11:47 server14 kernel: [6011663.543745] wanpipe1: Global Chip Configuration: used=1
Re: [asterisk-users] Using files .call or AMI
How would be the dialplan for this context from-lan ??? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sat, 12 Feb 2011 23:20:11 + From: ro...@firedrake.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? I'm using this to place calls from local IP-phones over the PSTN. So my script will generate, say: Channel: SIP/lanphone Context: from-lan Extension: 08001234567 taking the 0800... from the list of customer details. SIP/lanphone is the ID of the originating phone. Extension is the sequence the agent would dial if he were placing the call himself. The originating phone rings; when it's picked up, the Asterisk server calls the Extension number and bridges the two calls, so the local agent hears ringing tones from the far end. All the agent has to do is pick up the phone when it rings and put it down when the call is over. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On-Hold Music
On Friday 11 February 2011 16:37:49 Danny Nicholas wrote: Hi gang, In 500 words or less (if possible), please explain what is a legal music-on-hold file? My boss hates the stuff provided with the distribution and I figure that I'm asking for trouble if I take my Les Mis tracks and run them through Audacity and SOX to make new files. The proper licensing authority in the United States for hold music is BMI (Broadcast Music Inc). If you use music for MOH which is not royalty- free, then BMI requires a payment for each trunk line per year which is using such music. If you want to use for-royalty music, it is very possible, but it will be a continual expense. Not paying the fees upfront will cost you dearly in legal fees at the point at which you are caught (it's really only a matter of time). http://www.bmi.com/licensing/entry/534929 For future reference, I now work for a company which gets paid with fees generated by the music business (including MOH), so fair warning: if you announce that you're illegally evading such royalties (note that the use of royalty-free music, as is distributed with Asterisk, is perfectly legal), you may get a visit from the BMI enforcement division shortly thereafter. -- Tilghman -- _ -- 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] IP ban list by country
Hi everyone, I know it's off topic from Asterisk directly but yet related. What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. Thanks -- _ -- 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] IP ban list by country
On Mon, 14 Feb 2011, Bruce B wrote: What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. I compiled this list a few (6?) months ago by typing class A address blocks into Arin.net's 'whois' web page and noting which Regional Internet Registry it was allocated to. http://www.voip-info.org/wiki/view/allocated-class-a-ip-address-blocks After plonking this into a couple of production hosts, attacks of all ports dropped dramatically. I note there have been changes since then (128.0.0.0 was assigned to RIPE back in November), so if anybody wants to 'refresh' and post changes, please do. -- 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 -- 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] IP ban list by country
One possible advantage of the fact that IANA has depleted its pool of /8's (class A) is that if you are only filtering at that level the data is static now. It should never change again for IPV4. John On 2/13/2011 11:54 PM, Steve Edwards wrote: On Mon, 14 Feb 2011, Bruce B wrote: What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. I compiled this list a few (6?) months ago by typing class A address blocks into Arin.net's 'whois' web page and noting which Regional Internet Registry it was allocated to. http://www.voip-info.org/wiki/view/allocated-class-a-ip-address-blocks After plonking this into a couple of production hosts, attacks of all ports dropped dramatically. I note there have been changes since then (128.0.0.0 was assigned to RIPE back in November), so if anybody wants to 'refresh' and post changes, please do. -- _ -- 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] IP ban list by country
On Sun, 2011-02-13 at 22:54 -0800, Steve Edwards wrote: On Mon, 14 Feb 2011, Bruce B wrote: What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. I compiled this list a few (6?) months ago by typing class A address blocks into Arin.net's 'whois' web page and noting which Regional Internet Registry it was allocated to. http://www.voip-info.org/wiki/view/allocated-class-a-ip-address-blocks After plonking this into a couple of production hosts, attacks of all ports dropped dramatically. I note there have been changes since then (128.0.0.0 was assigned to RIPE back in November), so if anybody wants to 'refresh' and post changes, please do. Look at geoip and maxmind. Has a netfilter module to look up and pass/block based on geo-location via the registry information. Databases are available by subscription (fine grained, up to date) and a more general one for free use. see http://people.netfilter.org/peejix/geoip/howto/geoip-HOWTO.html Its been awhile since Ive used it and had to drop it because I needed access from the problem areas :( - but it worked very well at the time. BillK -- _ -- 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
