Re: [asterisk-users] GXP1400
Just a note to say that updating the phone to the latest beta code - 1.0.3.30 - mostly solved the problems I was having. It now gives ring back tones when making a local call however the early dial feature still does not work as it should and did in earlier (GXP280) phones. When making a call and hitting the digits the invite to Asterisk should be sent at the release of the key. In the GXP1400 it is sent about 100ms after the key is hit. Thus when you select a key you hear a short 100ms or so of the tone and the invite is sent to Asterisk even though you have not released the key. It does not sound natural and is not the way humans are use to interacting with a phone. I am hoping that Grandstream corrects this on a future firmware release. On Thu, 12 Apr 2012, Doug Crompton wrote: Some more input on this. I tried adding an explicit 'r' option to the dial command with no change. I also tried the 'progressinband=yes' setting in the section for this phone in sip.conf. No difference. No call tones and no or little blip DTMF tones in the handset when making a call. Offsite calls return call ring tone. Doug On Thu, 12 Apr 2012, Doug Crompton wrote: It is Asterisk 1.2 - yes I know old but it works for my application. The problem would not be gain, I have plenty of handset gain. It is the fact that the ring confirmation tones on local calls, which would be generated by asterisk are not there. They work fine on my two other older Grandstream phones. Also when I enter digits I do not hear the DTMF tones or only a very short blip. It completes calls fine. I have tried with and without 'early call' and bot 'in band' and 'RFC2833' DTMF. No change with either. Any ideas? Doug On Thu, 12 Apr 2012, Danny Nicholas wrote: You don't state which version of Asterisk you have this hooked up to, but dependent on that you might have rxgain and txgain available to you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Crompton Sent: Thursday, April 12, 2012 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] GXP1400 Just installed a Grandstream GXP1400 - I have several other older Grandstream phones installed. This one for some reason is operating differently. It works but I get no audible ring confirmation in the headset when I make a local call and I also only hear a blip of the DTMF ddigits or nothing at all in the headset as I key them in. Everything works OK, calls etc. just the audible feedback. Using an old Asterisk but all the other phones have worked fine. Is there some setting in Asterisk or the Grandstream I should be looking at to correct this? Doug -- _ -- 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 Although it is not true that all conservatives are stupid people, it is true that most stupid people are conservative * Doug Crompton * * Richboro, PA 18954 * * 215-355-5307* * * * d...@crompton.com* * http://www.crompton.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 Although it is not true that all conservatives are stupid people, it is true that most stupid people are conservative * Doug Crompton * * Richboro, PA 18954* * 215-355-5307 * ** * d...@crompton.com* * http://www.crompton.com * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
[asterisk-users] GXP1400
Just installed a Grandstream GXP1400 - I have several other older Grandstream phones installed. This one for some reason is operating differently. It works but I get no audible ring confirmation in the headset when I make a local call and I also only hear a blip of the DTMF ddigits or nothing at all in the headset as I key them in. Everything works OK, calls etc. just the audible feedback. Using an old Asterisk but all the other phones have worked fine. Is there some setting in Asterisk or the Grandstream I should be looking at to correct this? Doug -- _ -- 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] GXP1400
It is Asterisk 1.2 - yes I know old but it works for my application. The problem would not be gain, I have plenty of handset gain. It is the fact that the ring confirmation tones on local calls, which would be generated by asterisk are not there. They work fine on my two other older Grandstream phones. Also when I enter digits I do not hear the DTMF tones or only a very short blip. It completes calls fine. I have tried with and without 'early call' and bot 'in band' and 'RFC2833' DTMF. No change with either. Any ideas? Doug On Thu, 12 Apr 2012, Danny Nicholas wrote: You don't state which version of Asterisk you have this hooked up to, but dependent on that you might have rxgain and txgain available to you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Crompton Sent: Thursday, April 12, 2012 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] GXP1400 Just installed a Grandstream GXP1400 - I have several other older Grandstream phones installed. This one for some reason is operating differently. It works but I get no audible ring confirmation in the headset when I make a local call and I also only hear a blip of the DTMF ddigits or nothing at all in the headset as I key them in. Everything works OK, calls etc. just the audible feedback. Using an old Asterisk but all the other phones have worked fine. Is there some setting in Asterisk or the Grandstream I should be looking at to correct this? Doug -- _ -- 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 Although it is not true that all conservatives are stupid people, it is true that most stupid people are conservative * Doug Crompton * * Richboro, PA 18954 * * 215-355-5307* * * * d...@crompton.com* * http://www.crompton.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] GXP1400
Some more input on this. I tried adding an explicit 'r' option to the dial command with no change. I also tried the 'progressinband=yes' setting in the section for this phone in sip.conf. No difference. No call tones and no or little blip DTMF tones in the handset when making a call. Offsite calls return call ring tone. Doug On Thu, 12 Apr 2012, Doug Crompton wrote: It is Asterisk 1.2 - yes I know old but it works for my application. The problem would not be gain, I have plenty of handset gain. It is the fact that the ring confirmation tones on local calls, which would be generated by asterisk are not there. They work fine on my two other older Grandstream phones. Also when I enter digits I do not hear the DTMF tones or only a very short blip. It completes calls fine. I have tried with and without 'early call' and bot 'in band' and 'RFC2833' DTMF. No change with either. Any ideas? Doug On Thu, 12 Apr 2012, Danny Nicholas wrote: You don't state which version of Asterisk you have this hooked up to, but dependent on that you might have rxgain and txgain available to you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Crompton Sent: Thursday, April 12, 2012 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] GXP1400 Just installed a Grandstream GXP1400 - I have several other older Grandstream phones installed. This one for some reason is operating differently. It works but I get no audible ring confirmation in the headset when I make a local call and I also only hear a blip of the DTMF ddigits or nothing at all in the headset as I key them in. Everything works OK, calls etc. just the audible feedback. Using an old Asterisk but all the other phones have worked fine. Is there some setting in Asterisk or the Grandstream I should be looking at to correct this? Doug -- _ -- 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 Although it is not true that all conservatives are stupid people, it is true that most stupid people are conservative * Doug Crompton * * Richboro, PA 18954* * 215-355-5307 * ** * d...@crompton.com* * http://www.crompton.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 Although it is not true that all conservatives are stupid people, it is true that most stupid people are conservative * Doug Crompton * * Richboro, PA 18954 * * 215-355-5307* * * * d...@crompton.com* * http://www.crompton.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
[asterisk-users] Featuremap help
Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a feature that I want to direct to an extension in extension.conf Extension is 521 In features.conf - [applicationmap] dumpcaller = #9,callee,goto(521|1) show features - Dynamic Feature Default Current --- --- --- dumpcallerno def #9 Result -- Feature Found: dumpcaller exten: dumpcaller Dec 1 20:08:02 WARNING[18659]: res_features.c:958 feature_exec_app: Could not find application (goto(521|1)) I have tried many variations per docs. It shows goto(521|1) also tried (521,1) and dial(local/521/n) and many others. Always get could not find application. Any ideas? Doug ___ -- 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] Featuremap help
Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a feature that I want to direct to an extension in extension.conf Extension is 521 In features.conf - [applicationmap] dumpcaller = #9,callee,goto(521|1) show features - Dynamic Feature Default Current --- --- --- dumpcallerno def #9 Result -- Feature Found: dumpcaller exten: dumpcaller Dec 1 20:08:02 WARNING[18659]: res_features.c:958 feature_exec_app: Could not find application (goto(521|1)) I have tried many variations per docs. It shows goto(521|1) also tried (521,1) and dial(local/521/n) ad many others. Always get could not find application. Any ideas? Doug ___ -- 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] Choppy audio
OK in my research here is what I found. I seem to get the idea from what I read that ztdummy is not needed for 1.2 (and above) versions for anything but meetme. I never used ztdummy in my old system and it worked just fine. I really see no difference here with it running or not. Please confirm - is this timing required for just simple wav/gsm playback - like voicmail, etc. ? I played around with kernel options echo 0 /sys/devices/system/cpu/cpu1/online echo performance /sys/devices/system/cpu/cpu0/cpufreq/scaling_governor Which essentially makes it a 1 CPU system at max performance. NO CHANGE. Then put it back... echo ondemand /sys/devices/system/cpu/cpu0/cpufreq/scaling_governor echo 1 /sys/devices/system/cpu/cpu1/online Tried this echo 1 /sys/module/processor/parameters/max_cstate Default was 8. NO CHANGE. Let me reconfirm what I am hearing. The audio choppiness is subtle but definitely there. It seems to happen at the exact SAME place everytime I play it, which is suspicious! Could this be sometihng completely different then what we are suspecting? It seems to get out of sync. sometimes it seems it is playing future audio on top of current by only Ms's or maybe it is putting a hole there. Hard to tell. OH I also tried compiling with the O2 optimization rather then the O8 in the default Asterisk Makefile. Again NO CHANGE. Any more ideas would be helpful. Doug On Tue, 1 Jul 2008, Benjamin Jacob wrote: modprobe zaptel; modprobe ztdummy That will start zaptel and ztdummy after the 'zaptel stop'. Then restart asterisk. --- On Wed, 7/2/08, Doug Crompton [EMAIL PROTECTED] wrote: From: Doug Crompton [EMAIL PROTECTED] Subject: Re: [asterisk-users] Choppy audio To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, July 2, 2008, 1:58 AM OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start ___ -- 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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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] Choppy audio
Tzafrir, I have neither of those commands available here. Did a search to see if they were somewhere else but nothing. Using SUSE 10.2. In /proc I only have apci directory. Doug On Wed, 2 Jul 2008, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 02:47:34PM -0500, spectro wrote: On Tue, Jul 1, 2008 at 2:15 AM, Doug Crompton [EMAIL PROTECTED] wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I am getting choppy audio in voicemail and general message playback. see if disabling APM in your kernel solves the issue, add apm=off to kernel boot options. Anything still uses APM (as oposed to ACPI) noawadays? $ /sbin/acpi_available echo yes yes $ /sbin/apm_available echo yes (nothing) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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] Choppy audio
I am not sure who all see's this list but I do have a few questions that probably only the developers or somone really in the know of Asterisk could answer. - What is the requirement for timing vs. audio playback in Asterisk. Specifically voicemail and IVR's (Not meetme) - Has this requirment changed in newer versions? This obviously is when using Asterisk with no internal cards. I used Asterisk for several years with a P3 Linux system, NO timing, and it worked flawlessly. Now with this new Pentium Dual core system I do not have the perfect audio I experienced with the less powerful system. I fully know there are MANY variable here. It could be a combination of many things, including the OS (Linux Kernel) etc. BUT I offer this input, Music on Hold works fine. This uses mpg123. So why can this palyback fine and the other wav/gsm audio be choppy? I would gladly switch to a newer Asterisk (using 1.2.29) if someone said this was solved in that version. My system can obviously play (mpg123 - background) audio fine. Why then does Asterisk internal audio not also play well? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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] Choppy audio
Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I just switched over to this system from an older SUSE 2.4.10 kernel system. I am getting choppy audio in voicemail and general message playback. I installed Zaptel and ztdummy module and the following is zaptel status: slate:/etc/init.d # cat /proc/zaptel/1 Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 Is this indicating proper installation? Is there anything else I should try/do?? The choppyness is not extreme, just not perfect. I had no problem in my old system with 2.4. I had not even installed zaptel or ztdummy there. Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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] Choppy audio
As an addendum to my original message... In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, Doug On Tue, 1 Jul 2008, Doug Crompton wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I just switched over to this system from an older SUSE 2.4.10 kernel system. I am getting choppy audio in voicemail and general message playback. I installed Zaptel and ztdummy module and the following is zaptel status: slate:/etc/init.d # cat /proc/zaptel/1 Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 Is this indicating proper installation? Is there anything else I should try/do?? The choppyness is not extreme, just not perfect. I had no problem in my old system with 2.4. I had not even installed zaptel or ztdummy there. Doug * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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] Choppy audio
I saw that bug. Most of my files are WAV though. Would it apply to them also? Doug On Tue, 1 Jul 2008, Noah Miller wrote: Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- 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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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] Choppy audio
OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start If it is a relatively slow box, try getting the exact sound files you will be playing back, if you have the space (make menuselect). -bk ___ -- 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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- 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] CallWithUs Service?
John, I have used callwithus for almost 9 months. I left Gizmo when they had a doubling of their rates. I probably would have put up with the doubling but the fact that you can set your callerid at callwithus (and not Gizmo) was a big selling point. I have kept a minimum Verizon analog line for local and 911 dialing and I wanted my announced callerid to be my verizon number. With callwithus this is easy. I created a perl script to route the calls wither local to verizon or all else to callwithus. I also opted for the higher quality option. This is something like 1.5 rather than 1 cent/minute. This is not advertised but if you email them they will bump it up for you. I was not happy, at times, with the call quality at the 1 cent rate, although this may differ for you. I connect via iax, but you can also use sip. As far as I am concerned this is a no frills service that has worked for me. Doug On Thu, 13 Sep 2007, John Meksavan wrote: Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - robo dialer
Can anyone suggest a source for a free robot dialer or examples? I need to do some non-commercial auto dialing using Asterisk. Simple phone numbers in a file, line by line format. I found one called AstAutoDiaker but I was not able to get it to work and it appears to not be supported - no email response from author. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - robo dialer
Thanks Chris, Looks like a lot of capability. I can deal with Perl but I still was hoping for something a little more turnkey. Doug On Fri, 4 May 2007, Chris Bennett wrote: Hi Doug, Can anyone suggest a source for a free robot dialer or examples? I need to do some non-commercial auto dialing using Asterisk. Simple phone numbers in a file, line by line format. I found one called AstAutoDiaker but I was not able to get it to work and it appears to not be supported - no email response from author. If you are comfortable with Perl (programming language), or have access to somebody who is, you could get something working with the CPAN module Net::SIP .. http://search.cpan.org/~sullr/Net-SIP-0.26/ Amongst other things, it can be a 'phone' with the module Net::SIP::Endpoint. http://search.cpan.org/~sullr/Net-SIP-0.26/lib/Net/SIP/Endpoint.pod I havn't used it myself as yet but intend to in the next few months. If your requirements are for unattended dialling then you might find this option more flexible since you can make it do exactly what you want. Good luck with it anyhow! :) Regards, Chris Bennett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Using Asterisk/callerid with pay as you go
Yes, thank you. I found one, callwithus which has excellant Asterisk support, IAX/SIP and the support actually answered in minutes! So far good connects (usig IAX) and good prices. Lets hope it stays that way. I wonder why more companies can't be like that. This callerID thing is stupid. If you can go to many companies and can set it then why don't all companies offer that feature? It certainly would be a customer draw. Doug On Tue, 13 Feb 2007, Dovid B wrote: If you asked this question on the biz list you would get a lot of people that will tell you that they offer services where you can set the caller ID to what ever you want. To name a few:: Nufone Teliax Voipjet - Original Message - From: Doug Crompton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 12, 2007 10:33 PM Subject: [asterisk-users] Using Asterisk/callerid with pay as you go VOIPproviders I am curious how others handle call out VOIP and callerid. I have found numerous providers that are cheap and seem to have good voice quality but offer no provisions for callerid. I find it almost useless to use call out when the receiving party gets some bogus callerid number that has no relation to my call. I understand the big thing is spoofing callerid but are there any companies that offer a means of qualifying callerid so it works right? Like it or not callerid is used heavily and without a proper return ID many callee's don't answer and if they tried to return the call they get no where. Seems like a big problem to me. Very aggrevating. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk/callerid with pay as you go VOIP providers
I am curious how others handle call out VOIP and callerid. I have found numerous providers that are cheap and seem to have good voice quality but offer no provisions for callerid. I find it almost useless to use call out when the receiving party gets some bogus callerid number that has no relation to my call. I understand the big thing is spoofing callerid but are there any companies that offer a means of qualifying callerid so it works right? Like it or not callerid is used heavily and without a proper return ID many callee's don't answer and if they tried to return the call they get no where. Seems like a big problem to me. Very aggrevating. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3K to SPA3K DTMF issue
As per my (numerous) prior statements on this subject Asterisk WILL NOT properly work with the spa-3000 DTMF in rfc2833. Use INBAND when dealing with Asterisk on both the FXO/FXS ports of the spa3k if you are dealing with Asterisk. This is a setting in BOTH sip.conf and spa3k pstn and line 1 tabs. Symptoms are no DTMF after call completion (voicemail from outside to fxo) and IVR attempts from FXS attached analog phones. Using INFO negates use of dtmf control functions on your fxs/fxo ports - transfer etc. - Take your pick of what is more important to you. There should really be a wiki on this! It gets asked often. I might qualify that this is an issue with 1.2.x (and probably earlier) - not sure if any fixes make this work or work better in 1.4. Fault (apparently) lies with both sipura(linksys) and digium. Since in this case you are connecting the spa3k's thru Asterisk this would apply. I have not tried connecting two spa3k's directly together via network to see if they play together in this regard. Doug On Wed, 24 Jan 2007, Mark Coccimiglio wrote: My experience has been to be consistant. The only time I have had problems with DTMF is when I am not using the same DTMF encoding technique on all hardware. Your choices are: INFO, RFC2833 or INBAND. Some equipment also has an AUTO option but I would not recomend it. Stick with INFO or rfc2833 and be consistant across the enterprise. Mark C IS Manager http://www.psh-inc.com [EMAIL PROTECTED] wrote: Hi all, Has anyone faced an issue when sending DTMF from FXS of one SPA3K to FXO of another SPA3K through asterisk? Im not able to send it properly. Wanna be sure if its an issue faced by all.. If you have a fix for it, pls guide me. Thanks Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dtmf tones and SIP
I aaume you are calling in on a PSTN line? If so what fxo are you using with Asterisk. Doug On Wed, 17 Jan 2007, Giuffredi wrote: Hi list, I tried to use DISA in order to get the line when I call with my mobile phone but the system doesn't recognise my DTMF tones when I call to a SIP trunk. Everything is working Ok if I use a ZAP Trunks. I tried to google to find a solution but I wasn't able to find any. Any idea? I'm using trixbox 2 with 4 SIP trunks and a Zaptel TDM 400 card. Bye Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
I use default values for both of those. The big thing is to call youself. Use a cell, call a phone on the FXS. Hit a key on the cell and listen on FXS for DTMF. Make changes, reboot, and repeat. Hearing is believing. It is so much easier! I think you will find the inband will work. Doug On Mon, 15 Jan 2007, Louis-David Mitterrand wrote: On Fri, Jan 12, 2007 at 02:33:54PM -0500, Doug Crompton wrote: I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Thanks again Doug for that detailed explanation. As for the DTMF playback level and DTMF playback length settings, what do you use? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
I am not sure what you are asking? The problem is that rfc2833 does not play well with the spa-3000 and Asterisk. I am not sure if it is limited to just the spa3k. There is a bug causing this that has been documented. Google spa3000 dtmf bug asterisk for more info. The bottom line is that you need to use sip info (inband dtmf) if you desire dtmf transfer to the other party after the call has completed. Such as you call a bank, or you call your Asterisk voicemail, or your door lock which is actuated by dtmf. If none of these are of interest and you would rather have the dtmf features of Asterisk, then use rfc2833. you can't have both! Doug On Mon, 15 Jan 2007, Julio Arruda wrote: Doug, You are saying that RFC2833 somehow doesn't work if you have the Asterisk AND at a distinct time (still within the same call), the callee to see the DTMF, correct ? Would this be in any case ? (meaning, if the voice path is going via the Asterisk or UA to UA directly ?) I've my spa3k right now somewhat far :-), and I can't test it, but you know by any chance if SIP INFO would suffer from the same curse :-) ? From my limited understand, a big difference in this case is that RFC2833 really is in the RTP stream, but is not voice payload, while with SIP INFO, is done 100% out-of-band. Doug Crompton wrote: I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Connect a real analog phone to spa3000 fxs. Call it from another source, when connected send DTMF tones from that source. You should hear at least 100ms or more of the tone. inband should work. I suspect you are using alaw or ulaw codecs. There is really no reason to use anything else. When it does not work you will hear nothing more then a click or an ocassional to short tone. Another thing to check is that you should not be using any transfer options in your dial statement (t or T or other special features. You really have to listen to this to check it and make changes. Be sure to restart both spa3000 and asterisk when you make changes. Otherwise you can get fooled. If you are making the call from the spa3000 fxo to fxs, you need to have inband in BOTH. This is a known bug in Asteriskspa3000 for dtmf. I think the problem is somewhat shared but improvements in 1.4 may gelp or fic the problem. I am using 1.2 so I cannot answer that. Basically when using the spa3000 you have to make the choice of wether you want to be able to use dtmf features (transfer etc.)OR have the capability to send DTMF to or from the caller or callee. you really can't have both. Thus inband vs. rfc2833. I chose inband so I can interact with called ivr's and call in from pstn and access my VM. Doug On Fri, 12 Jan 2007, Louis-David Mitterrand wrote: On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote: The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling. Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be the line1 tab on spa3000. This applies to the fxo (pstn) also if you are using it for such things as ivr's. Thanks for your suggestion. We tried that without success (using firmware 3.1.7(GWc)) Do you think an upgrade to 3.1.10 might be warranted? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
OK... I understand. As I remember I did try other methods like INFO. It has been awhile. I think INBAND is the only one that worked for me. Doug On Mon, 15 Jan 2007, Julio Arruda wrote: Doug Crompton wrote: I am not sure what you are asking? The problem is that rfc2833 does not play well with the spa-3000 and Asterisk. I am not sure if it is limited to just the spa3k. There is a bug causing this that has been documented. Google spa3000 dtmf bug asterisk for more info. The bottom line is that you need to use sip info (inband dtmf) if you desire dtmf transfer to the other party after the call has completed. Such as you call a bank, or you call your Asterisk voicemail, or your door lock which is actuated by dtmf. If none of these are of interest and you would rather have the dtmf features of Asterisk, then use rfc2833. you can't have both! SIP INFO is not the same as Inband DTMF, that is why I'm asking. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling. Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be the line1 tab on spa3000. This applies to the fxo (pstn) also if you are using it for such things as ivr's. Doug On Fri, 12 Jan 2007, Louis-David Mitterrand wrote: Hello, Before throwing in the towel with my Sipura 3000 has anyone had much success with that adapter connected to a door phone? In our setup a doorphone is connected to the SPA's fxs port. When a visitor rings, asterisk calls a group of Polycoms and the person who answers has to enter *1 to trigger the door opening. However it seems the SPA doesn't relay the DTMF's to the doorbell. Any suggestions more than welcome, thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone
I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Connect a real analog phone to spa3000 fxs. Call it from another source, when connected send DTMF tones from that source. You should hear at least 100ms or more of the tone. inband should work. I suspect you are using alaw or ulaw codecs. There is really no reason to use anything else. When it does not work you will hear nothing more then a click or an ocassional to short tone. Another thing to check is that you should not be using any transfer options in your dial statement (t or T or other special features. You really have to listen to this to check it and make changes. Be sure to restart both spa3000 and asterisk when you make changes. Otherwise you can get fooled. If you are making the call from the spa3000 fxo to fxs, you need to have inband in BOTH. This is a known bug in Asteriskspa3000 for dtmf. I think the problem is somewhat shared but improvements in 1.4 may gelp or fic the problem. I am using 1.2 so I cannot answer that. Basically when using the spa3000 you have to make the choice of wether you want to be able to use dtmf features (transfer etc.)OR have the capability to send DTMF to or from the caller or callee. you really can't have both. Thus inband vs. rfc2833. I chose inband so I can interact with called ivr's and call in from pstn and access my VM. Doug On Fri, 12 Jan 2007, Louis-David Mitterrand wrote: On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote: The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling. Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be the line1 tab on spa3000. This applies to the fxo (pstn) also if you are using it for such things as ivr's. Thanks for your suggestion. We tried that without success (using firmware 3.1.7(GWc)) Do you think an upgrade to 3.1.10 might be warranted? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get dialed numbers in AGI
I am not sure if this is what you meant in your query. Here is a Perl
script that checks a dialed numbers areacode/exchange and determines if it
is a local or long distance call. I then send it out either PSTN or SIP.
The dialed number is sent to the script which returns true or false.
There might be a better way to check the return code but I was never able
to get a straight answer on that and this works.
This would work similarly in a PHP script with appropriate changes.
#!/usr/bin/perl
#
# Perl Script to determine if a call is in the local calling area
# Doug Crompton - 12/2006
#
# agi-local.agi
#
# Example in extensions.conf -
#
# exten = _215NXX,1,AGI(agi-local.agi)
# exten = _215NXX,n,Gotoif($[ ${localcall} = 1 ]? 10:20)
# exten = _215NXX,10,Dial(SIP/[EMAIL PROTECTED],60,T)
Local Call
# exten = _215NXX,11,Macro(failann,${DIALSTATUS})
# exten = _215NXX,20,Dial(SIP/[EMAIL PROTECTED],120,T)
Non Local Call
# exten = _215NXX,21,Macro(failann,${DIALSTATUS})
#
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$|=1;
my $lc = 0;
my $ret = 0;
if (my $exten = $input{'extension'}) {
my $area_code = substr($exten,0,3);
if ($area_code =~ /215|267|445/ ) {
$AGI-say_digits($area_code);
my $a215 = ($area_code =~ /215/);
my $a267 = ($area_code =~ /267/);
my $a445 = ($area_code =~ /445/);
my $exchange = substr($exten,3,3);
# Philadelphia Zone 4
# 215
$lc = $lc || ($a215 ($exchange =~
/214|268|270|281|288|289|305|330|331|332|333|335|338|342|4
#267
$lc = $lc || ($a267 ($exchange =~
/340|341|343|344|345|348|350|351|407|579|672|731/));
#
# Bensalem - Eddington - Cornwell Heights
# 215
$lc = $lc || ($a215 ($exchange =~
/202|244|245|352|447|604|633|638|639|642|645|650|688|929/)
# 267
$lc = $lc || ($a267 ($exchange =~
/223|332|520|522|523|525|526|527|529|681|704|771/));
#
# Bethayres - Huntingdon Valley
# 215
$lc = $lc || ($a215 ($exchange =~
/344|544|914|938|947|974|975/));
# 267
$lc = $lc || ($a267 ($exchange =~
/277|502|571|706|722|723|725|727|728|729/));
#
# Churchville - Feasterville
# 215
$lc = ($a215 ($exchange =~
/322|354|355|357|364|396|436|485|494|526|791|876|942|953/));
# 267
$lc = $lc || ($a267 ($exchange =~
/288|442|574|632|684|699|762|912|982|983|984|986|988|989|9
#
# Hatboro
# 215
$lc = $lc || ($a215 ($exchange =~
/259|293|315|323|325|328|347|385|394|420|441|442|443|444|6
# 267
$lc = $lc || ($a267 ($exchange =~
/220|280|282|317|387|532|537|615|732|803|960|961|963|965|9
#
# Langhorne
# 215
$lc = $lc || ($a215 ($exchange =~
/359|375|478|539|702|710|741|750|752|757|809|891|970/));
# 267
$lc = $lc || ($a267 ($exchange =~
/212|276|560|563|564|565|567|568|569|572|689|802|819|852/)
#
# Warrington
# 215
$lc = $lc || ($a215 ($exchange =~
/318|343|488|491|792|798|918/));
# 267
$lc = $lc || ($a267 ($exchange =~
/480|482|483|485|486|487|488|489|561|855|915|927/));
#
# Willow Grove
# 215
$lc = $lc || ($a215 ($exchange =~
/346|366|392|395|449|657|658|659|706|784|830|882|902/));
# 267
$lc = $lc || ($a267 ($exchange =~
/495|518|607|715|781|851|913|942|943|944|947|948|949/));
#
# Newtown
# 215
$lc = $lc || ($a215 ($exchange =~
/434|497|504|550|579|860|867|944|968/));
# 267
$lc = $lc || ($a267 ($exchange =~
/291|352|364|685|750|751|753|755|756|757|759/));
#
# Wycombe
# 215
$lc = $lc || ($a215 ($exchange =~ /598/));
# 267
$lc = $lc || ($a267 ($exchange =~
/396|491|493|494|719/));
#
if ($lc) {
$ret = 1;
$AGI-say_digits($exchange);
}
}
}
$AGI-set_variable('localcall',$ret);
exit;
___
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Re: [asterisk-users] Where is this hilarious Allison Smith file? (Also: Interview with Allison)
And the big question is are you going to use it in your home or a business. No one is going to sue an individual for using it for non-business even though it technically would be in violation. Same goes for MOH content. Doug On Thu, 11 Jan 2007, Paul wrote: [EMAIL PROTECTED] wrote: On Wed, 10 Jan 2007, Kevin P. Fleming wrote: Jerry Glomph Black wrote: I cannot find this file anywhere, despite thorough searching. Certainly not in the two usual big sound tarfiles. I have a great place for this file in my extensions.conf, no doubt. It has not been made available for distribution, sorry. Well, seeing its at the start of the interview, I think you'll find that it has been distributed ... Yes but 2 important questions arise: Was the interview was put under a license that allows redistribution? If so, does that license allow redistribution in part? If the answer to both is yes you can take your favorite sound editor to the file and create prompt files. I seriously doubt it was distributed under such terms. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk build for Suse 10.1
Formated your hardisk... wow that is nasty, but I also cannot understand how that could ever happen. There must be some other HW problem going on or you got a hold of some really bad code. What code (source or binary) and what were you doing when that happenned? Doug On Wed, 10 Jan 2007, Robert A. Rawlinson wrote: Thanks for the help. I was concerned because I tried once before and it formatted my hard disk. I wanted to be sure that did not happen again.\ Bob Rawlinson On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote: Has anyone heard of a build or instructions for installing Asterisk on a Suse 10.1 system? Bob Rawlinson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users k ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk build for Suse 10.1
With the RPM you get what you get. Why not get the latest source at digium and compile it. It is not hard to do. Doug On Wed, 10 Jan 2007, Robert A. Rawlinson wrote: Yes you are correct. I do NOT plan to use it again. I have downloaded the latest version and plan to do an install. I was hoping there might be an rpm for it but does not seem to be. Thanks all. Bob Rawlinson On Wed, 2007-01-10 at 18:25 +0100, Anton Frolov wrote: he is probably tried to install one of these All in one Asterisk CDs, that effectively formats the hard drive and installs everything from scratch, including the OS ;) And, yes, it will happen again, if he re-runs this CD... AF. Doug Crompton wrote: Formated your hardisk... wow that is nasty, but I also cannot understand how that could ever happen. There must be some other HW problem going on or you got a hold of some really bad code. What code (source or binary) and what were you doing when that happenned? Doug On Wed, 10 Jan 2007, Robert A. Rawlinson wrote: Thanks for the help. I was concerned because I tried once before and it formatted my hard disk. I wanted to be sure that did not happen again.\ Bob Rawlinson On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote: Has anyone heard of a build or instructions for installing Asterisk on a Suse 10.1 system? Bob Rawlinson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users k ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk build for Suse 10.1
When you say build do you mean a plug and play binary? I use SUSE 7.3 here and it is easy to get the source files and compile it. It should just work. The instructions would be in the README or INSTALL file in the source. 1. Get the source at digium (the 1.2.x version might be better to start with. 2. tar -xvzf version_name 3. cd to the directory tree made by step 2 4. Read the README and/or INSTALL text files for info on how to proceed 5. make 6. make install These steps would vary depending on wether you need zap or other drivers which would be compiled first. you would also want to download and install the sound files. There might be an easier install for a specific O/S version but I prefer to do things manually here. Of course I still do most everything at the command prompt also. I do not use a windowed system for my server. I don't see any reason why 10.1 would be any different then my 7.3 in the installation procedure. Doug On Tue, 9 Jan 2007, Robert A. Rawlinson wrote: Has anyone heard of a build or instructions for installing Asterisk on a Suse 10.1 system? Bob Rawlinson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.11 - ResponseTimeout being ignored
You need a 'waitexten()' after the background command.
On Tue, 9 Jan 2007, Erik Anderson wrote:
All - this is probably a simple problem, but I've been pulling my hair
out trying to figure out what I'm doing wrong. I'm building a
*simple* IVR menu. Here it is:
[main-menu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
exten = s,5,Background(logic-main)
exten = _4XX,1,Macro(stdexten,SIP/${EXTEN})
exten = 0,1,VoiceMail([EMAIL PROTECTED])
exten = 2,1,Directory(default|logic-boston]
exten = 2,2,Goto(main-menu,s,5)
exten = 3,1,Playback(logic-directions)
exten = 3,2,Goto(main-menu,s,5)
exten = t,1,GoTo(main-menu,s,5)
Everything is working fine except the ResponseTimeout(). My
understanding is that, as configured above, asterisk will wait for 30
seconds...if, after that amount of time, it hasn't received valid
digits, it'll jump to the t extension. That's not happening.
Immediately after the Background() sound file completes, I get this:
-- Playing 'logic-main' (language 'en')
== Auto fallthrough, channel 'SIP/445-0815e1d0' status is 'UNKNOWN'
Any ideas? This seemed like it should be simple, but it's getting the
best of me.
Thanks-
Erik
--
Erik Anderson
http://andersonfam.org
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* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
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RE: [asterisk-users] how to transfer calls when analog phone hasnotransfer button
Well it would be interesting to know what FXS device you are using to connect the analog phones. I use an SPA-3000 fxo/fxs and with it you could bypass Asterisk and connect the FXO to FXS or dial directly if it were so configured, so reinvite would work but wwould probably not be desired but that is not the question... I am using the SPA-3000 as both an FXO (connection to telco) and FXS (connection to my house analog phones) with Asterisk in between. I have said this before on here but I will say it again. With the SPA-3000 you cannot have analog phone feature keys, transfer etc. AND still be able to use DTMF for control outside of the dialplan. If you want feature key control then you would use rfc2833 DTMF, if you want to be able to use DTMF incoming or outgoing for control then you must use inband DTMF. It is either/or. My choice was to use inband and not have features selected for the analog phones. To often I would use these phines with banking or on incoming to control voicemail functions so I wanted that capability. In that case a hook flash works fine. If you have never done it just flash the hook for a second (or use the flash key on the phone) and you will get another dialtone. Then you can call another party, tell them you have a call to transfer and hangup or click again and bring them in as a conference. Doug On Fri, 5 Jan 2007, Don Pobanz wrote: Erick Perez Don, I suppose that in order for this to work i need canreinvite=no, right? No! It doesn't matter what you have for 'canreinvite' since 'canreinvite' is a SIP attribute, not an analog phone attribute. For analog phones, Asterisk will always be in the call path. :-) -- Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to email
There should be an example in your voicemail.conf Here is mine... mail is tagged from [EMAIL PROTECTED] and sent to [EMAIL PROTECTED] In voicemail.conf mailcmd=/usr/sbin/sendmail -f [EMAIL PROTECTED] [EMAIL PROTECTED] You of course would use the mailer that your system uses. I have sendmail on the same system as Asterisk. There are many other things you can define for mail but all should be in your example voicemail.conf Doug On Wed, 3 Jan 2007, Mark Greene wrote: Hey guys, I need to set up asterisk so that it sends the voicemail to the users email. I understand that I need to say attatch=yes, but what else needs to be done. I would think that somewhere I need to specify the server that it uses to send the email, etc. - Mark Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
try http://www.digitnetworks.com/X100P_FXO_PCI_Card_p/x100p.htm On Wed, 3 Jan 2007, Tzafrir Cohen wrote: On Tue, Jan 02, 2007 at 10:25:45PM -0500, cb wrote: On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. I just went thru this recently. I ended up buying a compatible modem on Ebay. You can find them easily if you search for FXO or X100 but then you may also end up paying a premium to get one that is specifically being sold to the Asterisk community. (keep in mind premium being around $30, so we still aren't talking about an outrageous price) Those 30$ cards are as good as the 10$ cards. Same low quality. They are nice for playing games. If you're lucky enough it may actually work for you. In the worst case you only lost 30$ ... What I did was checked the voip-info.org wiki on modem based FXOs and then searched ebay for modems listed with the correct chipsets. I lucked out and found one for $2.00 (with shipping I think it cost me $8.00 total). Mine is shows up as a Motorola X100 (or something to that effect). Seems to work fine, although I wasn't able to get Caller-ID working correctly (but I think that was a settings issue and I stopped pursuing it as it wasn't important for my pitching Asterisk). I don't recall any special issues with caller ID with X100P. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail and ip phones
What type of phone are you using? On my Grandstream 200 under the Account tab there is an item called voicemail ID This is the extension your would call to retrieve voicemail. In my case it is extension 80, so I have just 80 entered there. when I push the messages button on the phone it immediately connects me to voicemail for the extension I am calling from. You can set it up so all extensions are on the same voicemail or grouped according to your wishes. This is done in sip.conf (mailbox=) and in voicemail.conf to define mailboxes. Incoming messages for the associated mailbox will light the mesassge waiting indicator on the phone. Doug On Fri, 29 Dec 2006, Giedrius Augys wrote: Hi, In my ip phone is voicemail indicator, and also is a voicemail button (to access to voicemail server and ant to listen voicemail). My question is how to configure this button. In configuration I need to enter URL. What is the syntax of this URL, that IP Phone could fetch this voicemail from asterisk. Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Checking voicemail from outside
Or more likely the tone may not be getting to asterisk. What FXO are you using? External FXO's like the SPA3000 often need to be set to 'inband' DTMF - both in sip.config and in the device's config and be sure to restart Asterisk after doing this.. Easiest way to test this is to call yourself from your cell and see if you can hear the DTMF tones on the Asterisk side as you enter them on your cell. If you can not hear them then Asterisk won't decode them! This is also necesssary for outgoing FXO calls to enable use of external IVR's like banking and business voice menus. There is much about this on this list in the past and in Asterisk bug reports. It is not exactly clear where the problem lies but it appears to be a combination of Asterisk and the SPA3000. This might be fixed in version 1.4 but I have not heard any reports as yet. Doug On Thu, 28 Dec 2006, mitcheloc wrote: You could be using an older version of Asterisk that doesn't support it? On 12/28/06, Phil Finkler [EMAIL PROTECTED] wrote: Rob, Interestingly enough, I'm using that same sample macro, and that line is indeed in there, yet when I hit *, I hear the tone to leave a message. Any ideas? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE - Analog Phones with FSK/Stutter MWI
Not that I know of. I guess you could speed dial but then my Asterisk voicemail is 80 so how hard is it to pick up the phone and dial that. I never had phone company voicemail on a wired line so I don't know how that works but I suspect you have to dial your own 7 digit or 10 digit number??? Doug On Sat, 23 Dec 2006, Bob Chiodini wrote: Doug, Thanks for the info. I'm glad it works. One question: Is there some sort of one-button way to dial in to your voicemail? It seems I read something about it, when I was doing similar research? I think it was the Uniden CLX-465, which claims support of Phone Company voicemail. I could not find one locally, however. Happy Holidays Bob... Doug Crompton wrote: After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have phone company compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk and my SPA-3000. When there is a message waiting both the LCD display and a flashing indicator on the phone alert you. This is true for all extensions on the system, up to 8. These work with both FSK and Stutter tone. I did not turn on the tone MWI as the FSK worked fine. KX-TG5776S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with a 1.5 Full-Color (65k Color Capable) Backlit LCD on Handset $119.95 KX-TG5771S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with Talking Caller ID $99.95 KX-TG5622M 5.8 GHz FHSS GigaRange Dual-Handset Phone System $89.95 KX-TG5761S 5.8 GHz FHSS GigaRange Expandable Digital Cordless Phone with Talking Caller ID $89.95 In order for the external MWI to work you must turn on the message indicator and for units that have answering machines the machine must be turned off. Perhaps we could put together a list of analog phones that have this feature. I have been told that both Uniden and ATT have models that work but I have no knowledge of all that do in their entire line. Each brand has their own features and while the Panasonic is solid - I had a 2.4G system for years and really liked it - the Unidens seems to have more for the money but in this case not MWI. I guess you could tell I really wanted this MWI to work! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialplans for Asterisk?
Wow what a mess! I can imagine how much easier it would be if the world adopted a country/area/exchange scheme like in the US with known length. It must be complicated in Germany just within the country. At least in the US we know what the length should be so if we don't have that we know the number is in error. Doug On Fri, 22 Dec 2006, Anselm Martin Hoffmeister wrote: Am Freitag, den 22.12.2006, 00:53 -0500 schrieb Doug Crompton: Question... What is the purpose of the + before the number? Does anyone actually have to enter it? If so how would you do it? It is not used in the US but do I see it come in on SIP lines CID. I assume the CID ignores it in the number as I do not see it on the display. It is however stored in asterisk and when doing CID comparisions it can be a problem. The + is replaced by the telco you are connected to - by whatever the local prefix for international call is. In the US and Canada it will be 011, in most parts of the world 00, and there is Russia with its exotic 08 wait for beep 10... The + should work in GSM mobile networks and most SIP providers seem to accept it. For callerid, there seem to be several cases. One of my providers (the others manage better and always give 00492281234567 formatted numbers) gives CID as +491601234567 for calls from one German mobile network, 491637654321 from a second network and 02281234567 from landline, so my dialplan has to cope with that such that my endpoints show the proper number. This is done by the following logic: If number begins with +, strip it. If number begins with anything but 0, prepend 00. If number begins with 0049, replace by 0. Although in Germany you can dial 0049 (region) (number), readability is better when there is only the 0 (region) (number) on the display - especially as numbers tend to get long, and e.g. Grandstream BT-100 only have a 12-digit display. BTW the longest number I _think_ is planned in Germany is 9 digits after the area code for 2- and 3-digit area codes, 8 for 4-, and 7 for 5-digit areacodes. There is one exception though that I know of: One of our ministeries has usually 55- numbers (55 being their number, then four digits DDI), but their fax numbers are 8-digit. Thus resulting in total in 011-49-228-55-87654321 from US, 18 digits. If you can, leave room for long numbers. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
If this (or any) company is really stealing or not living up to a contract then why not report them as such, especially if they are US based. I would suspect you would have another route to take. If you don't do anything about it then they will just go on abusing others and getting away with it. At the very least the BBB (bbb.org) should be notified. They have a web site and if it is really wire/internet fraud then the FBI (www.fbi.gov/majcases/fraud/internetschemes.htm) has a site you can register a complaint with. Perhaps there should (or maybe there is) be a site that rates these companies. If someone really wants business then they will do their best to get on the top of the list. Doug On Fri, 22 Dec 2006, Kevin Walsh wrote: Andrew Joakimsen [EMAIL PROTECTED] wrote: NuFone isnt bad if you want a disposable termination account. But don't rely on it for anything. Well, the voice quality left a lot to be desired, so I didn't make a lot of use of the service anyway. Perhaps, if I had made more use of it, they wouldn't have been able to steal as much from me when I fired them for incompetence, laziness and general rudeness. Perhaps they think that if they rob enough people then they will be in a better position to pay their upstream provider - leading to less downtime and therefore less cause for customers to attempt to contact the idiot running their support department (Jeremy). They are in urgent need of a better business plan (one that doesn't rely upon raising money by stealing funds from customer accounts) and a complete change of staff. Oh, and a clue! -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
Well if this company is US based I would not think where you are matters if it is fraud. You could still enter a complaint at the FBI site. I would also think they would be working with the UK counterpart. Doug On Sat, 23 Dec 2006, Kevin Walsh wrote: Doug Crompton [EMAIL PROTECTED] wrote: If this (or any) company is really stealing or not living up to a contract then why not report them as such, especially if they are US based. I would suspect you would have another route to take. I am UK-based - there's not a lot I can do to a US-based company, and they know it. That is unless I'm willing to fly over there to sue them, which I'm not. In this case, it was better to just cut our losses and make sure as many others as possible know about our experience and are forewarned. I have already convinced several others to abandon NuFone. I was the one who recommended them in the first place, so I felt that I had to warn them and try my best to persuade them to migrate. Remember - sometimes, when you annoy one customer, you loose a lot more than just that one customer. Whatever playground victory NuFone thinks they won by helping themselves to the content of our account, they lost hundreds of times over in terms of the future revenue from the customers I know that they lost. If you don't do anything about it then they will just go on abusing others and getting away with it. That's why I feel it is right and proper to warn others. If NuFone carries on operating they way they do, they will lose a lot more than just the customers we had powers of persuasion over. The snowball effect induced by mounting bad publicity is a powerful thing, and not something any company wants to be on the receiving end of. I know that I have caused more damage this way than I could ever do by simply recovering the account balance in court, and I didn't have to fly anywhere to do it. NuFone's short-sighted and clearly criminal ways will come back to haunt them one day. At the very least the BBB (bbb.org) should be notified. They have a web site and if it is really wire/internet fraud then the FBI (www.fbi.gov/majcases/fraud/internetschemes.htm) has a site you can register a complaint with. I've never heard of the BBB. I have now - thanks. I doubt that NuFone's behaviour counts as fraud though. I'd class their actions as just plain old-fashioned theft. If you are a NuFone customer then I advise you to use up your balance and leave as soon as possible. It's very easy to do - especially if you're only using a company as an outgoing route and don't need to port a number to a new provider. If you know any NuFone customers then you should try to get them to do the same. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UPDATE - Analog Phones with FSK/Stutter MWI
After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have phone company compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk and my SPA-3000. When there is a message waiting both the LCD display and a flashing indicator on the phone alert you. This is true for all extensions on the system, up to 8. These work with both FSK and Stutter tone. I did not turn on the tone MWI as the FSK worked fine. KX-TG5776S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with a 1.5 Full-Color (65k Color Capable) Backlit LCD on Handset $119.95 KX-TG5771S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with Talking Caller ID $99.95 KX-TG5622M 5.8 GHz FHSS GigaRange Dual-Handset Phone System $89.95 KX-TG5761S 5.8 GHz FHSS GigaRange Expandable Digital Cordless Phone with Talking Caller ID $89.95 In order for the external MWI to work you must turn on the message indicator and for units that have answering machines the machine must be turned off. Perhaps we could put together a list of analog phones that have this feature. I have been told that both Uniden and ATT have models that work but I have no knowledge of all that do in their entire line. Each brand has their own features and while the Panasonic is solid - I had a 2.4G system for years and really liked it - the Unidens seems to have more for the money but in this case not MWI. I guess you could tell I really wanted this MWI to work! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Insert 1+areacode for VOIP calls
; Dial wether long distance is preceeded by 1 or not
; Dial LD via gizmo
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = _1NXXNXX,2,Macro(failann,${DIALSTATUS})
exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = _NXXNXX,2,Macro(failann,${DIALSTATUS})
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Re: [asterisk-users] International dialplans for Asterisk?
Question... What is the purpose of the + before the number? Does anyone actually have to enter it? If so how would you do it? It is not used in the US but do I see it come in on SIP lines CID. I assume the CID ignores it in the number as I do not see it on the display. It is however stored in asterisk and when doing CID comparisions it can be a problem. Doug On Fri, 22 Dec 2006, Michiel van Baak wrote: The above number looks like: +31318787243 Try to get that from your telco, it makes life way more easy. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
I haven't really been following this thread but doesn't the following
snipet kinda do this
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,playback(international-call)
exten = s,n,playback(please-enter-the)
exten = s,n,read(number,number)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,n,Macro(failann,${DIALSTATUS})
This matches 011 then could do any number of things. Here I just goto,
then it looks for more numbers (the announcement is optional) and then
dials them.
Maybe not what you are looking for but it is an example of Asterisk
matching an extension and then going on to take more digits that then
branch based on other digits. Here the 011 is prepended to the final
number.
BTW - what is a numer?
Doug
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Anthony,
Ok I understand. The 011 is unique though and I guess the problem is
the length of the remaining digits. This could vary based on country?? and
I suspect there is no unique rule that could be applied??? I have not
studied this but is there any uniqness to the remaining digits?
Doug
On Wed, 20 Dec 2006, Anthony Kepler wrote:
I have been using an approach such as this but am looking for something
else because of some limitations it has. The phone thinks it dialed,
and was connected to 011 (which it was)
As such, that will be stored in the phones dial history (redial if
nothing else).
I'm not even certain what I want is possible, which is why I'm asking
the list.
Thank you for your help once again though.
- Anthony Kepler
[EMAIL PROTECTED] | SIP/EMail
Doug Crompton wrote:
Well that is certainly an option but not all phones would have a send key
especially if you are using analog phones. I guess the # keys
functions in
that way on many of those.
I still like my wired phones to work like they use to. You dial a
number
and it executes the call immediately.
Ok I came up with one that I think would work, maybe needs some
refinement
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,read(number)
exten = s,2,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,3,Macro(failann,${DIALSTATUS})
This accepts the 011 prefix and then any number of following digits.
Terminator is timeout period OR # key to send. Change obviously for your
provider.
The read command has many options including saying a file. You could for
instance hear Country Code after dialing 011. This would clue you into
the fact that you were dialing and international call. There are also
digit limits and timeouts that can be set.
So if you use early dial this would be the only rule that would require a
wait or # key to send. I could certainly live with that.
Can anyone supply some international test numbers??? Say in the UK or
Germany or wherever outside the US.
Doug
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RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
On Wed, 20 Dec 2006, Michael Collins wrote: After listing all of that, then give us the description of what needs to happen next, the part about deciding which caller ID info to send. Pretend like you're explaining it to a bunch of idiots who understand only small words and short sentences. :) Damn, I didn't know Bush was subscribed to this list! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls disconnected after 1 hour
Sounds like a provider or equipment (FXO/FXS call timer) issue. What are the specs of your system? Doug On Thu, 21 Dec 2006, Klaverstyn, David C wrote: There seems to be something in Asterisk that disconnects the call at 1 hour. At 59 minutes there is a beep then 1 minute later the call is dropped. I have a basic install Asterisk Ver. 1.2.13. I have not specifically said that calls are to be disconnected at a certain time (not that I know how to do that). Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Early dial is a real nice feature BUT it requires that you carefully plan
and design your extensions. Each digit is accepeted by Asterisk and if a
match exists up to that point it will be accepted and dialed.
As an example, my internal extensions are 4xx and my internal special
extensions are 5xx. I chose those because they do not conflict with local
area codes or other first 3 digit sequences.
However if a call come in from, say, area code 512 (without the 1
prepended), and I have a local 512 extension, I would not be able to dial
that person back. It would instead go to the local 512, as this is
satisfied first.
Often callerID does not come in with the 1 before the area code. This is
what prompted me to put code in to append a 1 if none existed on the
incoming callerID. With the 1 appended there is no problem as 151 does not
match any local extension and I can use redial without problems.
Using 4 digit extensions would mostly eliminate this problem although you
still could not use 1xxx extensions.
Wildcard extension matches like X. or using the '.' anywhere in the
matches would not work.
You just have to use it and fix things as they come up. I think I have
most all cases trapped now!
Doug
On Tue, 19 Dec 2006, Anthony Kepler wrote:
Do you, Gordon or Doug, happen to place international calls with
early-dial enabled? What kind of extensions.conf magic do you work to
allow this?
I have been trying for some time to get this to work. (My message from
2006.11.03 regarding this is quoted just below)
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:I am trying to allow users to
place outgoing international calls from a
GXP-2000 with early dial enabled, connected to Asterisk 1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.
I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | SIP/Email
Gordon Henderson wrote:
On Sun, 5 Nov 2006, Doug Crompton wrote:
On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..
On the GPX2000's it's via the web interface under each of the 4 Line
configuration tabs. (so you'd have to set it on each account you were
using on the phone)
Gordon
Doug
On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:
Hi,
Where can I find that option?
Thanks
Jesus
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
On Wed, 1 Nov 2006, Henry.L.Coleman wrote:
I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?
Set the Early Dial option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the send button...
Gordon
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Re: [asterisk-users] Changing CALLERIDNUM on the fly
Thanks Anselm, That did it!
Doug
On Tue, 19 Dec 2006, Anselm Martin Hoffmeister wrote:
Am Dienstag, den 19.12.2006, 01:11 -0500 schrieb Doug Crompton:
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and this plays havoc with my redial if the 3 digit area
code matches a local 3 digit extension. All my outside calls are 10 digits
or 1+10 digits.
Doug
[from-pstn]
exten = s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
exten = s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
exten = s,3,set(__CALLERIDNUM=1${CALLERIDNUM}) if not add 1
exten = s,4,noop(${CALLERIDNUM}) and this still displays without
Replace CALLERIDNUM with CALLERID(num) on all occasions, and you will
not need underscores because this is a special variable anyway.
CALLERIDNUM is obsolete.
You could get along with one line less:
exten = s,1,GotoIf($[A${CALLERID(num):0:1} = A1]?3:2)
exten = s,2,Set(CALLERID(num)=1${CALLERID(num)})
exten = s,3,NOOP(Continue in Dialplan)
Note that my GotoIf contains the two additional A letters which is
important to avoid syntax errors if the CALLERID(num) is empty for
whatever reason. I do not know what ends up in your CALLERID(num) if the
number of the caller is not available (like anonymous or withheld) -
anyway, with this statement it will end up being prepended by 1. You
migth want to have a special case for that.
If your phones happen to also display CALLERID(name) you can use this to
lookup the phone number in a phone book (here in Germany there is an
online service for number reverse lookup which works for about 50% of my
callers) and set the variable.
BR
Anselm
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* 215-431-6307*
* *
* [EMAIL PROTECTED]*
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Sorry, I did not read the original message completely. The answer is no I do not make international calls. I do not know anyone in any other country to call! I do not have a rule for that but it should be easy to implement as 01x would not match anything I currently have for early dial. Would you always dial a 0 first for all international mumbers? Give me an example? Are you outside the US? If so give me your number and I will try it! Doug On Tue, 19 Dec 2006, Anthony Kepler wrote: I understand how early dial works (484 response and all that jazz), I also understand the NANP and how to keep my extensions from overlapping... but thank you for the tips. My question was: Do you place international calls from phones with early-dial enabled? If so, might you be willing to share the relevant portions of your dial plan that are concerned with placing said international calls? Thanks again, - Anthony Kepler [EMAIL PROTECTED] | SIP/Email ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Well that is certainly an option but not all phones would have a send key
especially if you are using analog phones. I guess the # keys functions in
that way on many of those.
I still like my wired phones to work like they use to. You dial a number
and it executes the call immediately.
Ok I came up with one that I think would work, maybe needs some
refinement
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,read(number)
exten = s,2,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,3,Macro(failann,${DIALSTATUS})
This accepts the 011 prefix and then any number of following digits.
Terminator is timeout period OR # key to send. Change obviously for your
provider.
The read command has many options including saying a file. You could for
instance hear Country Code after dialing 011. This would clue you into
the fact that you were dialing and international call. There are also
digit limits and timeouts that can be set.
So if you use early dial this would be the only rule that would require a
wait or # key to send. I could certainly live with that.
Can anyone supply some international test numbers??? Say in the UK or
Germany or wherever outside the US.
Doug
On Tue, 19 Dec 2006, Gordon Henderson wrote:
On Tue, 19 Dec 2006, Anthony Kepler wrote:
Do you, Gordon or Doug, happen to place international calls with early-dial
enabled? What kind of extensions.conf magic do you work to allow this?
I have been trying for some time to get this to work. (My message from
2006.11.03 regarding this is quoted just below)
Not me ( I'm in the UK FWIW).
I'm trying to get my users into thinking of the phones in the same terms
as they'd treat their mobiles - so get them to dial the full area code
starting with a zero (no 9 for outside line here, although I do support it
in addition to zero), and then pushing the send key after they have
entered the number... My reasoning for this is that it then mimics the way
they use their mobiles, (and who doesn't have a mobile these days?) and
you can dial the full number in the UK anyway without incuring any cost or
call routing issues (just time to dial the 4 or 5 digit prefix)
Gordon
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[asterisk-users] Changing CALLERIDNUM on the fly
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and this plays havoc with my redial if the 3 digit area
code matches a local 3 digit extension. All my outside calls are 10 digits
or 1+10 digits.
Doug
[from-pstn]
exten = s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
exten = s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
exten = s,3,set(__CALLERIDNUM=1${CALLERIDNUM}) if not add 1
exten = s,4,noop(${CALLERIDNUM}) and this still displays without
I tried no, one and two underscores with the CALLERIDNUM variable.
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Re: [asterisk-users] FYI Panasonic Wireless Phone MWI
Yes I was very specific. Go back to my original post - search Panasonic MWI - I described what I said and gave a link to the Panasonic specs for this phone which clearly states that the MWI light blinks with new messages and that phone company subscription to VM is required. I did not mention Asterisk because if it works with phone company VM it would work with Asterisk, assuming the FXO you were using was capable and setup correctly. Doug On Sat, 16 Dec 2006, Steve Prior wrote: Noah Miller wrote: Last week I asked about MWI indicators on wireless phones that would work with Asterisk. I sent a message off to Panasonic asking them about it because in their ads they specifically stated that the indicator works with and requires phone company voicemail subscription. That indicator will not work for your voicemail. We do not have any phone system that has a message alert indicator that will work both for your voicemail and your answering machine. How exactly did you phrase the question to their tech support? If you described Asterisk as an answering machine then you'd get the wrong answer. If you described Asterisk as a PBX which provides a signal just like a telco voicemail would, then the answer would make sense. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FYI Panasonic Wireless Phone MWI
Well it is not clear - their ads say one thing and they say another. At best there is confusion. If you really like the phone you could buy it somewhere where you could take it back and check it when you get it. Doug On Sat, 16 Dec 2006, Noah Miller wrote: Last week I asked about MWI indicators on wireless phones that would work with Asterisk. I sent a message off to Panasonic asking them about it because in their ads they specifically stated that the indicator works with and requires phone company voicemail subscription. That indicator will not work for your voicemail. We do not have any phone system that has a message alert indicator that will work both for your voicemail and your answering machine. Thanks Doug! That's good to know. I need to buy some analog cordless phones, and that one was on my list. Now I can scratch it off. If it helps, I have an ATT cordless that does have a working MWI indicator. The model is 9345. It's one of their very low-end basic models, so I'm guessing the better models would support MWI, too. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FYI Panasonic Wireless Phone MWI
Last week I asked about MWI indicators on wireless phones that would work with Asterisk. I sent a message off to Panasonic asking them about it because in their ads they specifically stated that the indicator works with and requires phone company voicemail subscription. The is the model TG5631. Specs here... http://www.amazon.com/Panasonic-KX-TG5631S-GigaRange-Cordless-Answering/dp/B000F4C2CA and this was the response Go figure! Dear MR CROMPTON Thank you for contacting Panasonic. The purpose of the message button of your phone system is to inform you that you have new messages in your answering system. That indicator will not work for your voicemail. We do not have any phone system that has a message alert indicator that will work both for your voicemail and your answering machine. We hope this information is useful to you. Thank You, Panasonic Consumer Support ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, Bluetooth, and wireless phone
With most of the new wireless phones now Bluetooth is anyone interfacing (pairing) them to Asterisk? It would be nice to just plop the phone down near the computer and have home phone access to it. I would be interested in hardware that might be used for this? I have a Bluetooth phone but not an interface for the computer. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone routing - curious what others are doing?
I just went through an exercise of writing a Perl script called from my Asterisk dialplan to look at a list of area codes and exchanges to determine which ones are local (no or little cost) under my current Verizon plan. I route calls outside of my local limits to Gizmo. It works fine but when I called Verizon to change (lower) my service it was a bewildering spider web of rates structures just in the Philadelphia metropolitan area. It made me wonder why I send any of my calls to Verizon! I was able to cut my Verizon cost down by about half. I wonder if any others are splitting calls like this or just biting the bullet and going 100% voip??? With Asterisk/Gizmo I have a local DID for $30/year plus I put $10 credit on callout last June and I still have $3 left. I prefer pay as you go rather than flat rate which at $20 or more a month would (for me) be a $150/year waste! When you have a Gizmo DID the callout CID is automatically the DID number. You can request a different number though as long as you have control of it. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5.8gig phone MWI
Well that is not exactly true. I nelieve it was clear as Steve stated below that his Uniden MWI does work with Asterisk. Many phones advertise compatibility with phone company MWI. Often people use phone company VM and then have stuttered dialtone as well as FSK signaling to tell the user there is a message waiting. There are definitely phones that will do this I am just trying to fine out which ones do and actually work! Asterisk through many FXS's would send this same signal. Doug On Fri, 8 Dec 2006, Tom Lynn wrote: You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. On 12/8/06, Doug Crompton [EMAIL PROTECTED] wrote: Thanks, but unfortunately that is an expensive 2 line phone compared to others in their line that have a base and two or three remotes for the same price. Seems a lot to pay for a MWI. I wonder if anyone has had experience with panasonic wireless 5.8gig and MWI?? They advertise compatibility on some models but I also saw a review comment that it did not work. Doug On Fri, 8 Dec 2006, Steve Prior wrote: Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5.8gig phone MWI
The ad at this Panasonic site is what is so confusing. They mention a MWI on the phone, which has a digital answering machine as part of the system. Then in the text they mention the MWI requires phone company subscription to voice mail. http://www2.panasonic.com/webapp/wcs/stores/servlet/vModelDetail?storeId=15001catalogId=13401itemId=96903catGroupId=25039modelNo=KX-TG5671S Doug On Sat, 9 Dec 2006, Steve Prior wrote: Tom Lynn wrote: You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. You're giving out totally incorrect information. The TRU-8866 unit I mentioned is a 2 line unit (which I wanted), but does NOT have a built in answering machine (which I didn't want). Uniden seems to offer models with and without answering machine function. However, despite the fact that it does not have a built in answering machine, the handsets and base unit both support MWI. I believe that Uniden does make a single line version of this phone, and I bet it also supports MWI - especially since they've standardized their handsets to be universal across their latest line. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 5.8gig phone MWI
Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5.8gig phone MWI
Thanks, but unfortunately that is an expensive 2 line phone compared to others in their line that have a base and two or three remotes for the same price. Seems a lot to pay for a MWI. I wonder if anyone has had experience with panasonic wireless 5.8gig and MWI?? They advertise compatibility on some models but I also saw a review comment that it did not work. Doug On Fri, 8 Dec 2006, Steve Prior wrote: Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Doug On Thu, 7 Dec 2006, John Marvin wrote: I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
I suggest you get the code I mentioned in my last message - it is c/c++ code and as is usually the case with Linux, all the source code is there. Looking at examples is a great way to learn. Doug On Tue, 5 Dec 2006, Zeeshan Zakaria wrote: What skills are needed to write a code yourself for X10, RS-485 or RS-232. I am planning to learn some programming so I can do the stuff myself which others haven't done yet. I once knew C/C++, and other electronic stuff, but because of not using it for years, revise and update them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it to a spare serial port on my linux server (asterisk resides there) and implemented with some mods the code mentioned earlier http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4beta3 help
No, no menuslect on system beside * I unzipped it, ran configure, then make (or make menuselect) they both give the same immediate error 3. From what I see with 1.4.x it might be good to have a completely seperste list. I suspect there will be tons of email volume once it's use or attempt of use ramps up! Doug On Fri, 1 Dec 2006, Tim Panton wrote: On 1 Dec 2006, at 03:49, Doug Crompton wrote: no - make menuselect - does the same thing. Have you got a (non asterisk) binary or shell script called menuselect in your path? try which menuselect Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] spa3k dtmf problem asterisk 1.2.x
Anyone that uses the spa3k with Asterisk knows about the dtmf issues of not being able to get tones properly to an IVR after call completion. You can make it work by eliminating ALL special keys - transfer, etc. in the dial and using inband signaling. This has been beat to death over the last year. My question is that there were patches to rtp.c that were an attempt to correct. I tried a few to no avail. Does anyone have a patch that works? I am currently using 1.2.13 My understand from googling this is that the problem is both a Sipura and Asterisk problem, although more of the blame is put on Asterisk. Also the rtp in 1.4 has been completely reworked. Has anyone tested this with the spa3k? Unfortunately 1.4 is a significant change that involves a great deal of time to test and is not at all like doing an upgrade within 1.2. So I am not inclined to go that route yet unless it fixes this problem. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendation for FXO
I have an spa3000 and in general it is OK. I have the echo at acceptable levels and often non-existent. The big grip is the failed rfc-2833. BUT it might be more the fault of Asterisk - See http://forum.voxilla.com/linksys-sipura-spa-users-group/dtmf-rfc2833-incompatibility-between-spa3000-asterisk-12306.html and my previous (today) message to this list on spa3k So other (external) devices may also not play well because of this problem. Digium it seems is not real excited to do anything about this as it is not an issue with their internal hardware and also 1.2.x is not on the front burner with 1.4 out. 1.4 wwith it's complete rewrite of rtp code might solve this issue. It is not clear as I have not seen an answer to my previous question on that. See... http://www.voipsupply.com/index.php?cPath=96 for a listing of the external ata's. Grandstream has one, as does linksys and others. I think they all have some issue. It is just picking the one that has the least or the most bearable for you. Unfortunately this may not become apparent until you get it and use it. I too would be interested in trying another fxo/fxs as my only experience is with Sipura. Doug On Fri, 1 Dec 2006, Martin Joseph wrote: Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased the wellgate 3701a, which was much better, but lacked ANY support from the manufacturer, and had some other semi-minor problems (rfc2833 didn't work, never got caller id working, etc.). Now the power problems seems to have done something bad to the PSTN gateway, as it appears to be up and running, but the gains are really whacked, and it's almost impossible to conduct a call through at this point. I tried hooking a handset directly to the PSTN line and that sounds fine. So, I would like to purchase another PSTN gateway which WORKS WELL with asterisk. I need it to hook up via ethernet, since my platform of choice (mac OSX) has no PCI card support. I only have one PSTN line, and already have other ATA's for FXSs, so I really only need one FXO port, although I realize there is no such animal. Any positive experiences with FXO gateways that connect via ethernet? Especially with a long loop/echo issues (ie not SPA3000)? Thanks in advance. Marty PS I am ready to spend to buy something quality. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4beta3 help
Ok Mine is probably old and not the right version. locate ncurses /lib/libncurses.so.4 /lib/libncurses.so.4.2 /lib/libncurses.so.5 /lib/libncurses.so.5.2 /usr/include/ncurses.h /usr/include/ncurses_dll.h /usr/lib/libncurses++.a /usr/lib/libncurses.a /usr/lib/libncurses.so /usr/lib/libncurses.so.1.9 /usr/lib/libncurses.so.1.9.7a /usr/lib/libncurses.so.2.1 as I am using a SUSE 7.3 system. It is time for a system rebuild and upgrade and I will probably wait until then to upgrade to 1.4 once there is an official release. Just playing now! Doug On Fri, 1 Dec 2006, Anthony Rodgers wrote: IIRC, menuselect requires ncurses-devel (or your distro's equivalent). CP On Dec 1, 2006, at 7:05 AM, Doug Crompton wrote: No, no menuslect on system beside * I unzipped it, ran configure, then make (or make menuselect) they both give the same immediate error 3. From what I see with 1.4.x? it might be good to have a completely seperste list. I suspect there will be tons of email volume once it's use or attempt of use ramps up! Doug On Fri, 1 Dec 2006, Tim Panton wrote: On 1 Dec 2006, at 03:49, Doug Crompton wrote: no - make menuselect -? does the same thing. Have you got a (non asterisk) binary or shell script called menuselect in your path? try which menuselect Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ??? http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety ?deserve neither liberty nor safety.? -- Ben Franklin (1759) *? Doug Crompton??? ?? * *? Richboro, PA 18954?? ?? * *? 215-431-6307 ??? ?? * *?? ??? ? ? ?? * * [EMAIL PROTECTED] * * http://www.crompton.com? * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ?? http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4beta3 help
I am running an old SUSE 7.3 system, 2.4 kernel and glibc 2.2 I picked up the ncurses-devel rpm and it now requires glibc 2.3 I found a glibc 2.4 rpm but I am a little reluctent to install it. It would be a disaster to lose this system. Are there any incompatibilities to look out for in installing glibc? In particuliar is there a kernel/glibc kernel match. Is the latest glibc backward compatible? I guess there could be a gcc problem starting at some rev. Doug On Fri, 1 Dec 2006, Anthony Rodgers wrote: IIRC, menuselect requires ncurses-devel (or your distro's equivalent). CP On Dec 1, 2006, at 7:05 AM, Doug Crompton wrote: No, no menuslect on system beside * I unzipped it, ran configure, then make (or make menuselect) they both give the same immediate error 3. From what I see with 1.4.x? it might be good to have a completely seperste list. I suspect there will be tons of email volume once it's use or attempt of use ramps up! Doug On Fri, 1 Dec 2006, Tim Panton wrote: On 1 Dec 2006, at 03:49, Doug Crompton wrote: no - make menuselect -? does the same thing. Have you got a (non asterisk) binary or shell script called menuselect in your path? try which menuselect Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ??? http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety ?deserve neither liberty nor safety.? -- Ben Franklin (1759) *? Doug Crompton??? ?? * *? Richboro, PA 18954?? ?? * *? 215-431-6307 ??? ?? * *?? ??? ? ? ?? * * [EMAIL PROTECTED] * * http://www.crompton.com? * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ?? http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2nd attempt - Return code - How to?
Can anyone give me some insight on this? If I am not making myself clear
please let me know.
At voip-info.org they show the following example
exten = s,1,Set(foo=${STAT(s,/var/t3)})
which I guess is suppose to work and make foo = size of t3
I did the following
exten = 542,1,Set(s1=${STAT(e,/var/lib/asterisk/t1)})
which should set s1 = 1 if the file exists and 0 if not.
but I get
Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not
registered
-- Executing Set(SIP/grandstream406-22e9, s1=0) in new stack
and in general I am confused about return codes. How would you use a
return code from the following
exten = s,1,System(somescript arg1 arg2)
Can someone give me a working example??? I keep getting the above error
Doug
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[asterisk-users] 1.4beta3 help
I do a ./configure successfully but when I try doing a 'make' I get error 1 - menuselect What am I doing wrong? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4beta3 help
no - make menuselect - does the same thing. On Thu, 30 Nov 2006, Matt Gibson wrote: Maybe you must run make menuselect before running make? Matt G On 30/11/06, Doug Crompton [EMAIL PROTECTED] wrote: I do a ./configure successfully but when I try doing a 'make' I get error 1 - menuselect What am I doing wrong? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Return code - How to?
At voip-info.org they show the following example
exten = s,1,Set(foo=${STAT(s,/var/t3)})
which I guess is suppose to work and make foo = size of t3
I did the following
exten = 542,1,Set(s1=${STAT(e,/var/lib/asterisk/t1)})
which should set s1 = 1 if the file exists and 0 if not.
but I get
Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not
registered
-- Executing Set(SIP/grandstream406-22e9, s1=0) in new stack
and in general I am confused about return codes. How would you use a
return code from the following
exten = s,1,System(somescript arg1 arg2)
Can someone give me a working example??? I keep getting the above error
Doug
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[asterisk-users] Return codes
How does one process a return code in Asterisk?
Example...
exten = s,n,Playback(/tmp/podcast/${CALLERIDNUM})
exten = s,n,System(rm /tmp/podcast/${CALLERIDNUM}.gsm)
If the caller hangs up on the playback command the file remove System
statement after it never gets executed. The playback command returns a -1
in this case and logs a warning. The only thing mentioned in the command
reference is 101 for file not found.
Doug
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Re: [asterisk-users] calls hang up even after Background() message eventhough response timeout is set to 10 sec
I had the same problem and found I needed a (for you example)
exten = s,n,waitexten
after the last background. This is shown in many examples and in others it
is not. Very confusing but I think adding this will work for you.
Doug
On Mon, 27 Nov 2006, Jeronimo Romero wrote:
I'm experiencing a strange problem. My inbound calls are hanging up
right after Background() message even though response timeout is set to
10 sec.
[voicepulseincoming]
exten=_X.,1,Answer
exte=_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1)
exten=_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1)
exten=_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)
[after-business-hours]
exten=s,1,Answer
exten=s,n,Set(TIMEOUT(digit)=10)
exten=s,n,Set(TIMEOUT(response)=10)
exten=s,n,SetVar(CALLFILENAME=${TIMESTAMP}:${CALLERIDNUM})
exten=s,n,Monitor(gsm,/var/spool/asterisk/monitor/${CALLFILENAME},m)
exten=s,n,Background(outside-business-hours)
exten=s,n,Background(main-auto-attendant)
exten=i,1,Goto(after-business-hours,s,7)
exten= 411,1,Directory(default)
exten= a,1,Goto(after-business-hours,s,7)
exten= o,1,Goto(after-business-hours,s,7)
The call hangs up without respecting the 10 second response timeout.
I've seen people posting this issue but I haven't seen the solution.
Any help would be greatly appreciated.
The asterisk console spits out the following message:
-- Playing 'outside-business-hours' (language 'en')
-- Executing BackGround(IAX2/voicepulse01-1,
main-auto-attendant) in new stack
-- Playing 'main-auto-attendant' (language 'en')
== Auto fallthrough, channel 'IAX2/voicepulse01-1' status is 'UNKNOWN'
-- Hungup 'IAX2/voicepulse01-1'
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Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
This is my spa3k fxs port sip.conf params. This uses the default context
in my extensions.conf
What are you having trouble doing? Can you make calls out to PSTN? Is it
just incoming call that are not ringing?
Doug
[sipurafxs1]
type=friend
regexten=405
username=sipurafxs1
secret=
context=default
context=from-pstn
callerid=Doug Crompton 405
host=dynamic
nat=no
port=5061
canreinvite=no
disallow=all
allow=alaw
allow=ulaw^M
allow=gsm
allow=g723.1^M
[EMAIL PROTECTED]
dtmfmode=rfc2833
On Sat, 18 Nov 2006, Larry Alkoff wrote:
Doug Crompton wrote:
Doug, please forgive me but I'm still having trouble understanding two
points from your last response.
Can you please post your extension 405 (analog extension on spa3k) in
sip.conf
and your [sipurafxs1] ?
I finally understand that INRINGSDEV is meant to specify which analog
and SIP phones to ring at extension INRINGSEXT = 405 and would like to
see just how you do it.
Larry
On Wed, 15 Nov 2006, Larry Alkoff wrote:
Thank you very much Doug for your detailed response to my question.
I'm working on a new sip.conf and extensions.conf using your code as a
guide.
Questions:
In INRINGSDEV what does sipurafxs1 and grandstream406 refer to?
The comment says ring analog phones on spa3k fxs but grandstream406
seems to refer a Grandstream sip phone, not an analog one.
Does INRINGSDEV mean ring a specific sip phone and the analog ones?
INRINGSDEV is a list of the devices you want to ring when you use this
variable in the dial statement. sipurafxs1 is the fxs side of the spa3k
and I have one grandstream 200, at extension 406, named grandstream406.
The analog extension, fxs on the spa3k, is 405.
How would I ring all the _sip_ phones when a pstn call comes in?
My macro 'ring-all' ?
You just add them all together in the ring statement with the as in my
INRINGSDEV variable. Actually the use of the variable was taken from
sample code given to me when I started out. It is probably a good idea
though. you could just put them all in the dial statement but if you use
it in more than one place it is handy to just change it in one place and
use the variable.
SIP/sipurafxs1SIP/grandstream406thirdfourth.
Notes:
Your sipurafxo1 is my spa3k-pstn-in defined in both Sipura and sip.conf.
My extension to ring incoming calls is 120 vs your 405. All ok on these
two.
I'm nearly there thanks to you.
OK glad it helped. If you have any other questions let me know. The spa3k
has a million settings.
Larry
Doug Crompton wrote:
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set to inband here and in spa3k to fix a problem with DTMF not
working for menus from PSTN. This was discussed earlier and is a problem
in asterisk that may (or may not) be solved in 1.4. I am using earlier
version. Inband must also be specifed in spa3k pstn.
[sipurafxo1]
type=peer
username=sipurafxo1
secret=x
canreinvite=no
context=from-pstn
host=dynamic
nat=no
port=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
In extensions.conf. This is a little fancy but the bottom line is that it
ends up in either a day or night mode. Only day shown. The spa3k fxo in
sip calls the from-pstn but the pstn-day-time (below) could be relabeled
from-pstn to always go to phones. The night mode basically goes to VM.
INRINGSEXT and INRINGSDEV are just variables defined to -
INRINGSDEV=SIP/sipurafxs1SIP/grandstream406 ; ring analog phones on spa3k
fxs
INRINGSEXT=405 ; the extension to ring for incomming calls
The stdexten macro is just the standard one in sample extension file.
[from-pstn]
exten = s,1,GotoIf($[ ${day-night} = 0 ]?2:10
exten = s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
exten = s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
exten = s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten = s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},)
On Tue, 14 Nov 2006, Larry Alkoff wrote:
My SIP phones can dial out through Sipura SPA3k to POTS for local and
911 calls _but_ incoming POTS calls are being swallowup somehow.
Am I on the right track with the code snippit below?
sip.conf:
-
In sip.conf the following code is _supposed_ to ring the SIP phones when
a POTS line call comes in through Sipuara to Asterisk.
[spa3k-pstn-in] ; Pots-line-in from Sipura
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
host=dynamic
type=friend ; should be peer if incoming only ??
context=[macro-ringall] ;ring all the sip phones
secret=x
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
On Wed, 15 Nov 2006, Larry Alkoff wrote:
Thank you very much Doug for your detailed response to my question.
I'm working on a new sip.conf and extensions.conf using your code as a
guide.
Questions:
In INRINGSDEV what does sipurafxs1 and grandstream406 refer to?
The comment says ring analog phones on spa3k fxs but grandstream406
seems to refer a Grandstream sip phone, not an analog one.
Does INRINGSDEV mean ring a specific sip phone and the analog ones?
INRINGSDEV is a list of the devices you want to ring when you use this
variable in the dial statement. sipurafxs1 is the fxs side of the spa3k
and I have one grandstream 200, at extension 406, named grandstream406.
The analog extension, fxs on the spa3k, is 405.
How would I ring all the _sip_ phones when a pstn call comes in?
My macro 'ring-all' ?
You just add them all together in the ring statement with the as in my
INRINGSDEV variable. Actually the use of the variable was taken from
sample code given to me when I started out. It is probably a good idea
though. you could just put them all in the dial statement but if you use
it in more than one place it is handy to just change it in one place and
use the variable.
SIP/sipurafxs1SIP/grandstream406thirdfourth.
Notes:
Your sipurafxo1 is my spa3k-pstn-in defined in both Sipura and sip.conf.
My extension to ring incoming calls is 120 vs your 405. All ok on these
two.
I'm nearly there thanks to you.
OK glad it helped. If you have any other questions let me know. The spa3k
has a million settings.
Larry
Doug Crompton wrote:
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set to inband here and in spa3k to fix a problem with DTMF not
working for menus from PSTN. This was discussed earlier and is a problem
in asterisk that may (or may not) be solved in 1.4. I am using earlier
version. Inband must also be specifed in spa3k pstn.
[sipurafxo1]
type=peer
username=sipurafxo1
secret=x
canreinvite=no
context=from-pstn
host=dynamic
nat=no
port=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
In extensions.conf. This is a little fancy but the bottom line is that it
ends up in either a day or night mode. Only day shown. The spa3k fxo in
sip calls the from-pstn but the pstn-day-time (below) could be relabeled
from-pstn to always go to phones. The night mode basically goes to VM.
INRINGSEXT and INRINGSDEV are just variables defined to -
INRINGSDEV=SIP/sipurafxs1SIP/grandstream406 ; ring analog phones on spa3k
fxs
INRINGSEXT=405 ; the extension to ring for incomming calls
The stdexten macro is just the standard one in sample extension file.
[from-pstn]
exten = s,1,GotoIf($[ ${day-night} = 0 ]?2:10
exten = s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
exten = s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
exten = s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten = s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},)
On Tue, 14 Nov 2006, Larry Alkoff wrote:
My SIP phones can dial out through Sipura SPA3k to POTS for local and
911 calls _but_ incoming POTS calls are being swallowup somehow.
Am I on the right track with the code snippit below?
sip.conf:
-
In sip.conf the following code is _supposed_ to ring the SIP phones when
a POTS line call comes in through Sipuara to Asterisk.
[spa3k-pstn-in] ; Pots-line-in from Sipura
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
host=dynamic
type=friend; should be peer if incoming only ??
context=[macro-ringall];ring all the sip phones
secret=x
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
extensions.conf
context to ring all SIP phones when a POTS call comes into SPA3k:
[macro-ringall] ; ring all SIP phones
exten = s,1,Dial(SIP/120SIP/121SIP/122SIP/124SIP/125SIP/126SIP/127)
exten = s,2,hangup
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
___
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Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com
Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?
Well I have a Grandstream 200 in a home application and so far I have been happy with it. My biggest complaint is that 99% of these IP phones are black!! One of the reasons I bought the 200 was because it has a bright red, see across the room, message waiting indicator. I have not seen that spec'ed on other phones. That doe not meant they don't have it, it is just not spec'd. I imagine the multiline LCD's have it on the screen, but you would not see that unless you specifically walked over and looked. I would be interested if any other phones have message waiting indicators as visible as the GS 200. Doug On Wed, 15 Nov 2006, Tom Vile wrote: They brake easy. Speaker phone is not very good. Overall sound not good compared to a Snom, Polycom or Cisco phone. Drop registrations with Asterisk randomly. Power supplies die. Had 4 out of 10 go bad within a year. LCD backlight died on 2 that I deployed. We only do the Snom 320 or 360's now and are just as easy to configure and have alot of great options as well. On 11/15/06, Jeronimo Romero [EMAIL PROTECTED] wrote: We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF intercom right out of the box. They can also be centrally managed and provisioned. They also sound great and work in a very intuitive way. We don't have real life experience deploying this phone so I'm just going to ask: Is there a catch? Why the huge price difference? These phones seem to do everything a busy corporate office would need. Is there a big qualitative difference between this phone and Polycom501/601?? Is there a major problem with this phone not disclosed by the manufacturer or vendors. Some feedback from people who have deployed them would be great. Thanks In advance. JR ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?
Is that not what the early dial option is for? In the GS 200 it dials immediately based on * dialpans with this option set. I am sure the 2000 has it also. I have no wait for any valid dialplan. Doug On Thu, 16 Nov 2006, Henry.L.Coleman wrote: I have deployed the Grandstream 2000 with very little hardware problems. Early firmware was petty rough but from 1.1.1.9 onwards is very robust. Frankly it represents the best bang for your buck. The only thing that I would like to see is a dial plan (which would speed up dialing). Most IP-phones don't have this anyway so it's not a big deal. The only other IP-Phone that I would consider is the Aastra 480i which is of a higher overall quality but the display is not as bright as the GXP 2000 and is difficult to view. PS they haven't ironed out all the bugs with the sidecar (56 button BLF/DSS) Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Doug, Just a note on this subject: I have a Snom 320 at home, and it's got a nice orange MWI that's pretty visible (especially if the apartment is dark). At the office I have a Polycom 501. It's got a great red light right at the top of the phone in the middle. It's very visible unless the phone isn't facing you at all. Alex On 11/15/06, Doug Crompton [EMAIL PROTECTED] wrote: Well I have a Grandstream 200 in a home application and so far I have been happy with it. My biggest complaint is that 99% of these IP phones are black!! One of the reasons I bought the 200 was because it has a bright red, see across the room, message waiting indicator. I have not seen that spec'ed on other phones. That doe not meant they don't have it, it is just not spec'd. I imagine the multiline LCD's have it on the screen, but you would not see that unless you specifically walked over and looked. I would be interested if any other phones have message waiting indicators as visible as the GS 200. Doug On Wed, 15 Nov 2006, Tom Vile wrote: They brake easy. Speaker phone is not very good. Overall sound not good compared to a Snom, Polycom or Cisco phone. Drop registrations with Asterisk randomly. Power supplies die. Had 4 out of 10 go bad within a year. LCD backlight died on 2 that I deployed. We only do the Snom 320 or 360's now and are just as easy to configure and have alot of great options as well. On 11/15/06, Jeronimo Romero [EMAIL PROTECTED] wrote: We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF intercom right out of the box. They can also be centrally managed and provisioned. They also sound great and work in a very intuitive way. We don't have real life experience deploying this phone so I'm just going to ask: Is there a catch? Why the huge price difference? These phones seem to do everything a busy corporate office would need. Is there a big qualitative difference between this phone and Polycom501/601?? Is there a major problem with this phone not disclosed by the manufacturer or vendors. Some feedback from people who have deployed them would be great. Thanks In advance. JR ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice
[asterisk-users] 900 rules
I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 900 rules
Ok so ONLY 900 numbers are pay. Next question 18XX numbers. are they all toll free? Is there any space in 8xx that is used otherwise? Doug On Tue, 14 Nov 2006, Eric ManxPower Wieling wrote: Doug Crompton wrote: I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? _1900NXX _NXX976 ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set to inband here and in spa3k to fix a problem with DTMF not
working for menus from PSTN. This was discussed earlier and is a problem
in asterisk that may (or may not) be solved in 1.4. I am using earlier
version. Inband must also be specifed in spa3k pstn.
[sipurafxo1]
type=peer
username=sipurafxo1
secret=x
canreinvite=no
context=from-pstn
host=dynamic
nat=no
port=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
In extensions.conf. This is a little fancy but the bottom line is that it
ends up in either a day or night mode. Only day shown. The spa3k fxo in
sip calls the from-pstn but the pstn-day-time (below) could be relabeled
from-pstn to always go to phones. The night mode basically goes to VM.
INRINGSEXT and INRINGSDEV are just variables defined to -
INRINGSDEV=SIP/sipurafxs1SIP/grandstream406 ; ring analog phones on spa3k
fxs
INRINGSEXT=405 ; the extension to ring for incomming calls
The stdexten macro is just the standard one in sample extension file.
[from-pstn]
exten = s,1,GotoIf($[ ${day-night} = 0 ]?2:10
exten = s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
exten = s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
exten = s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten = s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},)
On Tue, 14 Nov 2006, Larry Alkoff wrote:
My SIP phones can dial out through Sipura SPA3k to POTS for local and
911 calls _but_ incoming POTS calls are being swallowup somehow.
Am I on the right track with the code snippit below?
sip.conf:
-
In sip.conf the following code is _supposed_ to ring the SIP phones when
a POTS line call comes in through Sipuara to Asterisk.
[spa3k-pstn-in] ; Pots-line-in from Sipura
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
host=dynamic
type=friend ; should be peer if incoming only ??
context=[macro-ringall] ;ring all the sip phones
secret=x
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
extensions.conf
context to ring all SIP phones when a POTS call comes into SPA3k:
[macro-ringall] ; ring all SIP phones
exten = s,1,Dial(SIP/120SIP/121SIP/122SIP/124SIP/125SIP/126SIP/127)
exten = s,2,hangup
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
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Re: [asterisk-users] Ringing phones
You did not mention what your FXO (connection to PSTN) hardware is??? Depending on what it is there may be configuration options for things like 'ring thru' and wether the fxo answers or passes the call to * Doug On Wed, 8 Nov 2006, Matt wrote: Hi, I have a system that connects to the PSTN.What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call? The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they have answered! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg errors? Other anomalies? Check those capacitors!
The motherboard's capacitor? What is that? Since there are probably a hundred or more caps on the MB, how did you determine that? Was it burned? Other than that, without making either capacitance or noise tests I can't imagine how you would make that assumption. Doug On Wed, 8 Nov 2006, Ronald Lewis wrote: Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did overlook one thing: * The motherboard's capacitor! Yep, you guessed it! It was bad. Now, I do not have any problems (I didn't bother replacing the motherboard, ended up using a spare PC). Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan Question
Thanks, that set off a light bulb In my spa3K my incoming dialplan was set to (S0:405) Since this is a one FXO unit and my [from-pstn] will always be that line can I make it generic and use the 's' extension as I described? If so what would that spa3k dialplan be? just s0 ? Doug On Tue, 7 Nov 2006, Anselm Martin Hoffmeister wrote: Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton: I am trying to do something that I see describe in a book and it is not working In my sip.conf, I have in my [fxo] context=from-pstn I then have in extensions.conf [from-pstn] exten s,1,answer() exten s,2,playback(blah) etc. It never answers but if I do this [from-pstn] exten _x.,1,answer() exten _x.,2,playback(blah) it works. Why does the 's' extension not work here? If fxo means your SIP provider, and you register with him, a specific extension will be called. Which one shall be called can be selected by the last parameter of the register statement, e.g. register = 075741:[EMAIL PROTECTED]:5060/492281234567 will cause the incoming calls to appear in extension 492281234567. Comes in handy if you have several accounts with a single SIP provider: This way, you can simply distinguish the outward phone number for which the call came in. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan Question
Answering my own question. If you want to connect an spa3K with generic pstn inbound do the following... for the pstn to voip dialplan in the pstn tab - (S0:ip-address-of-*) in sip.conf [sipurafxo] context=from-pstn etc. Then in * extensions.conf use the s extension. [from-pstn] exten = s,1,answer() exten = s,2,dial. etc. Makes it alot easier as you do not have to deal with extension matching when you know where it is coming from. Doug On Tue, 7 Nov 2006, Doug Crompton wrote: Thanks, that set off a light bulb In my spa3K my incoming dialplan was set to (S0:405) Since this is a one FXO unit and my [from-pstn] will always be that line can I make it generic and use the 's' extension as I described? If so what would that spa3k dialplan be? just s0 ? Doug On Tue, 7 Nov 2006, Anselm Martin Hoffmeister wrote: Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton: I am trying to do something that I see describe in a book and it is not working In my sip.conf, I have in my [fxo] context=from-pstn I then have in extensions.conf [from-pstn] exten s,1,answer() exten s,2,playback(blah) etc. It never answers but if I do this [from-pstn] exten _x.,1,answer() exten _x.,2,playback(blah) it works. Why does the 's' extension not work here? If fxo means your SIP provider, and you register with him, a specific extension will be called. Which one shall be called can be selected by the last parameter of the register statement, e.g. register = 075741:[EMAIL PROTECTED]:5060/492281234567 will cause the incoming calls to appear in extension 492281234567. Comes in handy if you have several accounts with a single SIP provider: This way, you can simply distinguish the outward phone number for which the call came in. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial plan Question
I am trying to do something that I see describe in a book and it is not working In my sip.conf, I have in my [fxo] context=from-pstn I then have in extensions.conf [from-pstn] exten s,1,answer() exten s,2,playback(blah) etc. It never answers but if I do this [from-pstn] exten _x.,1,answer() exten _x.,2,playback(blah) it works. Why does the 's' extension not work here? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
On the Budgetone 200 it is in the account tab settings of the web setup and it does work here with asterisk and my dialplans.. Doug On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote: Hi, Where can I find that option? Thanks Jesus -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Gordon Henderson Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m. Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones? On Wed, 1 Nov 2006, Henry.L.Coleman wrote: I came to the same conclusion. There is one thing however that the GXP2000 needs in my opinion. There is no dial plan avaiable in the configuration, this means that when dialing a number there is a slight delay before it actually dials. With a dial plan the dialed number is sent immeadiately the pattern is match ed so it saves a second or two. Maybe they will fix this? Set the Early Dial option - it's on a per-line basis, then as soon as Asterisk gets a number it can dial, it will. No need to wait the 4 seconds or press the send button... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3k wired to PAP2 for echo testing
Yes I agree, the SPA3000 can be a bear with echo on the PSTN. I did find that using older fimware helped some and that the levels - there are 4 settings - FXO/FXS in/out can be juggled to help. I also found out after adding a Budgetone 200 that I had much less echo problem going through it and the spa3000 FXO - vs. using the local analog phones on the spa3000 fxs port to FXO port. So some of the answer might be to get rid of as much (or all) local analog as you can. I plan to buy more hard sip phones and do that here eventually. This is ultimately more flexible as each extension has it's own number and they can dial each other as well as dial more then one place simutaneously. The big problem is that SIP phones are generally ugly and black and not styled for home use. Doug On Sun, 5 Nov 2006, James Harper wrote: In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset -- (FXS)SPA3000(FXO) -- PAP2 And set the Line1 dialplan on the SPA3k to '(:@gw0S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! If anyone on this list with a SPA3k (that doesn't have any local echo problems on the PSTN port) and an ATA with a FXS port, could they please try the above setup and post the results (including SPA3k hardware and firmware versions, and the ATA used)? I wonder if there is a problem with some versions of the SPA3k where there is some sort of inbalance on the PSTN port that causes echo right there rather than further down the line? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR problem
Can anyone tell me why the following code snipet does not behave the way I would expect? The background audio files are gsm and play fine. Here is what happens. When the set-day-night context is called it plays the menu asking to select 0,1, or 2. It then immediately falls through and terminates never waiting for the selection. Doesn't the timeout function determine the length of time it waits regardless of the actual sound file length? I have tried lengthening the time to no avail. The line marked below waitexten was added to make it work. It does not have the same functionality though with this added. The code minus this line is textbook basic IVR as far as I can tell. Comments??? Doug [set-day-night] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,Set(TIMEOUT(response)=10) exten = s,5,Background(doug/select-day-night) exten = s,6,waitexten added line exten = s,7,hangup() exten = 0,1,SetGlobalVar(day-night=0) exten = 0,2,Playback(doug/day-night-mode-reset) exten = 0,3,Hangup() exten = 1,1,SetGlobalVar(day-night=1) exten = 1,2,Playback(doug/day-mode) exten = 1,3,Hangup() exten = 2,1,SetGlobalVar(day-night=2) exten = 2,2,Playback(doug/night-mode) exten = 2,3,Hangup() exten = t,1,Goto(#,1) ; If they take too long, give up exten = i,1,Playback(invalid) ; That's not valid, try again exten = i,2,Hangup() ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR problem
Ok thanks... well my point was that numerous examples show NOT using waitexten but rather just background with the appropriate timers set. So in my example below s,6 and s,7 would be eliminated. This seems to not work though. As soon as the backgrouond message is complete the plan exits to nowhere rather than obey the timeout (t) plan. At least this is the way the book says it should be. Doug On Fri, 3 Nov 2006, Bruce Reeves wrote: I have a similar IVR and use WaitExten(5) to give 5 seconds for the extension to be entered. I have not tried using the Timeout options so I am not sure how they should affect your dialplan. On 11/3/06, Doug Crompton [EMAIL PROTECTED] wrote: Can anyone tell me why the following code snipet does not behave the way I would expect? The background audio files are gsm and play fine. Here is what happens. When the set-day-night context is called it plays the menu asking to select 0,1, or 2. It then immediately falls through and terminates never waiting for the selection. Doesn't the timeout function determine the length of time it waits regardless of the actual sound file length? I have tried lengthening the time to no avail. The line marked below waitexten was added to make it work. It does not have the same functionality though with this added. The code minus this line is textbook basic IVR as far as I can tell. Comments??? Doug [set-day-night] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,Set(TIMEOUT(response)=10) exten = s,5,Background(doug/select-day-night) exten = s,6,waitexten added line exten = s,7,hangup() exten = 0,1,SetGlobalVar(day-night=0) exten = 0,2,Playback(doug/day-night-mode-reset) exten = 0,3,Hangup() exten = 1,1,SetGlobalVar(day-night=1) exten = 1,2,Playback(doug/day-mode) exten = 1,3,Hangup() exten = 2,1,SetGlobalVar(day-night=2) exten = 2,2,Playback(doug/night-mode) exten = 2,3,Hangup() exten = t,1,Goto(#,1) ; If they take too long, give up exten = i,1,Playback(invalid) ; That's not valid, try again exten = i,2,Hangup() ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: My Phone Review- Large Scale Corp Deployment.
I am certainly not an expert on this but I bought a Budgetone 200 and for $59 in the VOIP market I think it is an excellent bargain. While not feature laden it does what it is suppose to. My biggest complaint is that most all of the VOIP phones are made for the business and not the home market. 99% of them are black and ugly and not something I would like in my kitchen or bedroom. I choose the Budgetone 200 for it's style as well as the mostly high ratings I have seen for it. It would be nice if it was offered in white or beige though. I also like the message waiting indicator that works well with * VM. It is easy to setup and easy to use. If you are thinking about using it in a larger scale you should buy one and evaluate it. No big loss if you don't like it and you should even be able to return it with most suppliers if you are not satisfied. It sounds as good or better than my analog system through an spa-3000 and appears to have less echo problems. Doug On Sat, 4 Nov 2006, Doug Meredith wrote: Eddie Johnson Jr [EMAIL PROTECTED] wrote: Did you test Snom or Sipura hard ip phones? I was considering Budgetone for an office of 10 users. After reading your testimonial I will have to re-think my selection. I don't know if you have used a BT but this isn't a phone that I would put in my garage much less my office. Poor sound quality, lousy display and a horrible interface. If you want something cheap to prove that VoIP actually works, then sure. If you plan to use it to talk to someone, then look elsewhere. Some other quick notes: Aastra 480i works pretty well. Sipura SPA-841 isn't much of a phone. Linksys SPA-921 941 work pretty well but suffer from occasional lockups and spontaneous reboots (latest firmware). Linksys WIP-300 is crap. Doug -- Doug Meredith 506-854-7997 ext. 801 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk
I am using an Sipura 3000 here and it is working (mostly) fine but I had a lot of learning to do in the beginning. I missed the original question and problem?? Perhaps you could state that again of refer me to it? Keep in mind there are two major firmware versions for the 3000. Version 2 and 3. In version 2 you have to do some extra stuff to get it to allow Asterisk to answer. In version 3 there is an option to allow it to pass the ringing call to Asterisk which makes it a lot easier. when doing this you need to set the delay such that at least 1-2 rings are received before you do this or the CID does not get passed properly. I have it set to 4 or 4 seconds as I recall. The version 3 firmware IS useaable on the version 2 hardware. I am using it here. I did find that an earlier version 3 firmware had better echo cancelation then the lastest version though. Actually the version 2 firmware was even better but lacking the option to have Asterisk answer the phone, so I opted for the compromise. I would be glad to send my settings it you like. My HW version is 2.0.1 and firmware is 3.1.3 The Sipura site only has the latest version 3 - 3.1.10 I believe. There is an (australian?) site that has all the past versions available for download. Google for it. Doug Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF and ivr systems
I am using an SPA-3000 and after using Ulaw (G711) on my * to FXO side (not FXS) it seems to work fine. I believe there is a problem with * and DTMF that is being worked on and there should be something new to (try) in the 1.4 release this summer. I don't think you mentioned your hardware. I think this works differently depending on the device. When I used RFC-2833 and AUTO on my SPA-3000 FXO side, I got the short muted blip of DTMF you mentioned. Changing to INBAND now gives at least a 250ms+ duration, which has been long enough to work with anything I have tried. The reason that * is checking/muting DTMF is that it has to look for transfer and other codes to process. So this is the other gotcha. You cannot use transfer or any other features because even with inband * seems to mute when using this - I.E. if your transfer feature is #9, when you hit the # key it will mute and the # key will never get through. I believe the correct behavior would be to wait and see if the second key was sent in a certain time and if not then send the orginal key. Rather tricky but it does not work in the current *. So I do not use any features that use the DTMF keys. So to summarize INBAND, Ulaw, on FXO and not use of DTMF features and it should work. Turning off feaures means no Tt, etc in dial command. I do transfers or parking, if needed via switch hook. Hopefully this will be fixed and I can go back to rfc-8322 and also have my features back in the future. It also really helps to call yourself, say to a cellphone, and listen for this behavior! Doug On Sat, 1 Jul 2006, Monty Lilburn wrote: Hi, I too am experiencing the same problem you have. I am using inband DTMF processing with ULaw (G711) and like you I notice that Asterisk seams to be passively listening to the line waiting to hear a DTMF. When it hears a DTMF it mutes the handset and regenerates my original DTMF (in a very short burst) which often gets missed by the remote party. This is especially true for IVR systems. I haven't come across a configuration option that keeps Asterisk from muting the handset and regenerating the original DTMF. Perhaps if Asterisk saw that the active channel was using inband processing with G711 it could leave everything alone and just let the user's dtmf go through unfettered! If this isn't possible for some technical reason I wonder if there is a configuration option that allows the user to set the duration of the regenerated dtmf? Maybe a developer will see this and can comment. Best regards, Monty On Thu, 29 Jun 2006, Shane wrote: Hello, Ther's probably a simple answer to this but I've searched around and haven't located anything as yet. Is there a way to have DTMF tones passed through Asterisk without it messing with them? I am using a tdm21b card and when I call an ivr system from the telephone handset (routed over sip or iax2) such as telebanking, the ivr has trouble recognizing tones. When I tested this with a remote party, I was told tones were breaking up. For example, a long press would result in a click, some silence and a small dtmf on the remote end. Triggering a speed dial didn't go well either as he heard only a few tones. I have dtmfmode=inband in sip.conf and have tried relaxdtmf=yes in zapata.conf. I realize Asterisk does need to detect dtmf for things like call parking but can it just pass the audio to the other side with no regard for whether it's dtmf digits? IE. long press results in long tone, etc. Best, Shane ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] best hardphone for Asterisk?
I guess I did not make my point clearly enough. I already do have just that. An spa-3000 with ALL internal analog phones on it's on FXO. But this gives just ONE extension for all phones. Yes I could get more FXS's and run seperate wires. So with that background what would be nice is a wireless device like the Panasonic cordless with one base and multiple phone capability that connected via ethernet and serves the phones. Just wishful thinking. I will stick with what I have until something useful, sylish, and less expensive arrives. Doug On Mon, 26 Jun 2006, Michael George wrote: On Mon, Jun 26, 2006 at 12:08:48AM -0400, Doug Crompton wrote: Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. 1. If you have *, you don't necessarily need multiple handsets off of one base. 2. Cordless phones also require power 3. If the multi-handset cordless phone does suit your needs best, then get a SIP ATA device like a Sipura or IAXy and you should have your needs met. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] best hardphone for Asterisk?
Iain, Thanks for the repsonse but you are kidding me right? From what I can see if I bought this phone and two remotes my outlay would be close to $800 US. This is NOT a home device unless you have nothing better to do with your money! You can buy a lot of single line wireless phones and FXS devices for that amount! Doug On Mon, 26 Jun 2006, Iain Barker wrote: Doug, What you are describing sounds like the Aastra 480-CT, a base Ethernet/SIP screenphone supporting multiple wireless handsets [but as this is a non-commercial list I won't go into more detail here, google for the above model number if you're interested in more info.] - Iain --- Message: 4 Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT) From: Doug Crompton [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] best hardphone for Asterisk? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. Doug Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] best hardphone for Asterisk?
Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. Doug On Sun, 25 Jun 2006, shadowym wrote: I believe all three Aastra phones(9112, 9133i, 480i)have exactly the same handsets and speakerphone hardware which is THE most important thing. After that it just depends on what additional features you want. They are ALL solid business phones IMHO and Aastra's support of the Asterisk community and the end user is outstanding! -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: Friday, June 23, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] best hardphone for Asterisk? On Fri, 2006-06-23 at 10:39 -0700, shadowym wrote: I love my Aastra 9133i with v1.4 firmware. Pretty much everything just works with Asterisk right out of the box and it has all the features I need. If cost is important the 9112i would be better. I install all three Aastra models and the sound quality is good across the range. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
Ok Now I understand. You mentioned you have an SPA-3000 in your inventory.
That is what I use here and I do not load or use zap or pri modules. I use
the 3000 as my fxo/fxs via sip on my local network. I have no cards in my
computer. You could do the same for testing of your problem.
Doug
On Tue, 20 Jun 2006, John Millican wrote:
Okay here goes,
I guess I misunderstood Doug's question about the far end interface. I have no
availability for high speed internet at my house to place a VoIP call over.
So, I have a standard phone plugged into the PAP2, The PAP2 plugs into the
network at my house to which the asterisk box is also connected, the
asterisk box has an FXO card that has the PSTN line plugged into it, this is
where the ZAP channel comes in. when i dial a local number asterisk simple
dials the number out the pstn line. If i dial a long distance number, the *
box dials a local phone number that I have through my VoIP provider which is
answered by an * box that I have at a different location using a line in
extensions.conf like:
Dial(zap/1/my_sip_numberww${EXTEN});
this way when the second * answers the phone it get the ${EXTEN} that I
actually dialed and dials it out over the cable connection. I hope i was a
little clearer this time and sorry for the confusion.
John M
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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Re: [Asterisk-Users] DTMF Talk off
John,
Well I have mentioned this before but not sure if I did to you... The
only was I was able to get things working well here was to use ulaw/alaw
only on the FXO and also use INBAND. This is specified on both the PSTN
tab of the 3000 and in the FXO sip.conf for it. I did not have to do this
for the fxs, line1 spa3000 port.
The other thing that is useful is to call yourself. If you have a
cellphone, call it and listen while hitting DTMF keys in both directions.
You should hear at least 500ms or more of the tone.
Doug
On Tue, 20 Jun 2006, John Millican wrote:
I have looked at that as a solution but haven't been able to get the dtmf to
work reliably. When I dial a local call i get connected okay and can
obviously connect to the second * box with out problem, the problem comes in
trying to get the ${EXTEN} portion of the dial string
Dial(zap/1/my_sip_numberww${EXTEN});
to the second * box. It tends to not see the DTMF correctly for the number I
want to call. When I watch this on the CLI through SSH on the second box i
see the call come in and go to the correct extension where it waits for
digits to dial and I get sporadic results, sometimes no digits are recognized
and sometimes 2 or 3, but never all correctly. I have tried increasing, and
decreasing the wait in the dial string in my home * with no luck. Any hints
on how to get the 3000 and the * box to talk better? If I could get this to
work through the 3000 I would be a very happy camper as it would open up some
possibilities that I can't do cost effectively otherwise. I will start to
route all local calls out the 3000 though for testing in the mean time.
Thanks for the ideas,
John M
On Tuesday June 20 2006 10:16 am, Doug Crompton wrote:
Ok Now I understand. You mentioned you have an SPA-3000 in your inventory.
That is what I use here and I do not load or use zap or pri modules. I use
the 3000 as my fxo/fxs via sip on my local network. I have no cards in my
computer. You could do the same for testing of your problem.
Doug
On Tue, 20 Jun 2006, John Millican wrote:
Okay here goes,
I guess I misunderstood Doug's question about the far end interface. I
have no availability for high speed internet at my house to place a VoIP
call over. So, I have a standard phone plugged into the PAP2, The PAP2
plugs into the network at my house to which the asterisk box is also
connected, the asterisk box has an FXO card that has the PSTN line
plugged into it, this is where the ZAP channel comes in. when i dial a
local number asterisk simple dials the number out the pstn line. If i
dial a long distance number, the * box dials a local phone number that I
have through my VoIP provider which is answered by an * box that I have
at a different location using a line in extensions.conf like:
Dial(zap/1/my_sip_numberww${EXTEN});
this way when the second * answers the phone it get the ${EXTEN} that I
actually dialed and dials it out over the cable connection. I hope i was
a little clearer this time and sorry for the confusion.
John M
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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Re: [Asterisk-Users] DTMF Talk off
Check
http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.html
On Sun, 18 Jun 2006, John Millican wrote:
Hello all,
I have seen some chatter again about DTMF. I see most of the talk about DTMF
around not being able to get an external IVR to recognize digits, not a big
issue for me at this time but sill interesting. My issue though, is with
talk off on a zap channel. It seems to be getting worse or maybe my patience
is thinning. All my calls go out and come in pstn through an FXO as I do not
have high speed available here at home. My Current setup is:
Phone--PAP2-- * ---PSTN---Voip number to * at another location(that has
high speed)---send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I also
remeber seing something about relaxdtmf being set to something other than yes
or no, so I looked in chan_zap.c and found relaxdtmf in many places but it
looked to my inexperienced eye that it could only be set to 'yes' or 'no',
but i did find some mention of tonlength (at line 10858)
followed that to zaptel.c (line 3357) where it said :
if ((tdp.dtmf_tonelen 4000 ) || (tdp.dtmf_tonelen 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values ignore it.
Any ideas what might happen if i increased the minimum time value? if this is
indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel = 1
same for channel 2
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten = s,1, NoOp(${CALLERID} time ${DATETIME});
exten = s,2, Dial(sip/677sip/666,30,tT);
exten = a bunch of stuff to do with agi look ups and voicemail
leave/retrieve
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
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