RE: [asterisk-users] OT: Sipura SPA-3000 ATA DirectingCalls to Asterisk
If you just save the page (from the browser) it will have the entire configuration asa continuous html file. NewSipuraUtil is good too (more as a backup and restore sort of facility). Bothways no passwords are passed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar Sent: Friday, July 07, 2006 6:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Sipura SPA-3000 ATA DirectingCalls to Asterisk On Thu, 06 Jul 2006 19:34:01 -0400, Brian Capouch [EMAIL PROTECTED] wrote: Thomas Kenyon wrote: For some reason when I do this, It only works if I have callerID switched off, otherwise I get authentication errors. Do you know of anyway to bulk-save the contents of all the config screens on that unit? Try NewSipuraUtil at http://www.dualarrow.com If so, I could scrub the passwords and send you the config for the one I'm using. I just checked; I am getting the CallerID just fine when I bring calls into my Asterisk box via the SPA3K. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [Asterisk-Users] [OT] Disconnect Tone in US
Have you compared the disconnect tone on both units - after all, you say one of them works fine. On mine (default since I didn't change this part) it is : [EMAIL PROTECTED],[EMAIL PROTECTED];4(.25/.25/1+2) on the PSTN page In case you're wondering, it is not working here either (not US).I use * # to force a hangup or *1# for a follow-on call (before the answering side hangs up). HW is 2.0.1(4e16) and firmware 3.1.7(GWg) - Original Message - From: Paul Dugas [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Thursday, May 18, 2006 05:20 Subject: [Asterisk-Users] [OT] Disconnect Tone in US I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in a Voxilla posting [1] to measure the disconnect tone and ensure the configs match. This seems totally off base to me but I figure I'd ask. Anybody care to comment? Do disconnect tones vary wildly in the US? Do they vary between two lines from the same telco switch? Paul ___ --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] Confused !
I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able to tell if this would work. Your restriction is what the service provider allows. Most (that I've used) allow g729. I know it uses more bandwidth than g723 but nothing like G711 (ulaw or alaw) and from my experience, the quality is quite reasonable. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: Re: [Asterisk-Users] Confused ! thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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 -- Got my Gmail ? ___ --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] Confused !
1. In the extension definition, insert canreinvite=yes for each of your clients. 2. In the trunk definition, insert canreinvite=yes Read about reinvite at http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite Apparently some hardware does not like it, and obviously, both the client and the service provider with have to be able to use the same codec (for them to be able to talk to each other) but better if Asterisk is restricted to that codec on both sides to start with. Please understand, I am trying to help and I don't know which parts (of what I'm saying) are not entirely accurate but normally if I say something wrong there are enough people who clamour to correct me. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 14:16 Subject: Re: [Asterisk-Users] Confused ! how to use reinvite in my asterisk setup ? thanks Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able to tell if this would work. Your restriction is what the service provider allows. Most (that I've used) allow g729. I know it uses more bandwidth than g723 but nothing like G711 (ulaw or alaw) and from my experience, the quality is quite reasonable. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: Re: [Asterisk-Users] Confused ! thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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] Confused !
Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager) v0.1.0 released
For a linux newb who needs to wake up ? How does one do this ? Copy/create a app_wakeme.c in the source directory then compile asterisk ? How do I call it in dialplan ? - Original Message - From: Michael Iedema [EMAIL PROTECTED] To: Asterisk Users asterisk-users@lists.digium.com Sent: Monday, May 08, 2006 14:47 Subject: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager) v0.1.0 released Greetings, I'm releasing my first attempt at Asterisk development in hopes of gathering a bit of feedback. It's just a wake-up call manager but, seeing as the only one I was able to find was a php-agi script, perhaps an actual app will be useful to someone. Anyway, here's the details. Features: * User can set / cancel wake-up calls for their extension * Option to snooze is presented during the actual wake-up call * Month / Year / Leap-Year changes are accounted for Todo: * Account for time-zone differences between caller and PBX * Account for daylight savings time changes * Abstract / redo the prompting to account for word order differences in different languages * Implement both new and old loaders...currently we're developing for trunk only !! As the last point implies, this first version is only intended for users of trunk Ok, enough show. The code is available from our freshly installed, still not perfect, website at: https://tk-labor.dyndns.org:8443/pub/?q=node/1 --Michael I. ___ --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] Voicemail indication for analog phones
There are analog phones with support for showing an envelope (or message) on receiving VMWI. DECT phones from Siemens and Panasonic do that too (although they are digital and wireless but they interface with analog lines). In fact, the Siemens phones have a function to tell how many messages are waiting. Let us not forget that Asterisk can also transmit an email message to advise (and deliver) voicemail. - Original Message - From: Garth Summey To: asterisk-users@lists.digium.com Sent: Sunday, May 07, 2006 23:46 Subject: [Asterisk-Users] Voicemail indication for analog phones I have 20 or so users with analog cordless phones connected via a 24port FXS box (vegastream). The vegastream supports voicemail indication via a studder tone, which is great but I have some users asking for a more positive (or proactive) indication of voicemail. I had a couple ideas; 1.If an extensionhas a voicemail then ring the extension once per hour for one ring, then hangup. 2.If an extension has a voicemail then ring the extension once per hour and if answered, play something along the lines of "You have X voicemails, please dial *97 to listen to your messages." Something that I have to keep in mind is that some of these phones are in residential areas, so either of the above actions would need to be time limited (active from 8:00am to 9:00pm for example). Has anyone done something like this before, anyone have any better ideas? Will this have to be executed by Cron (probably because the above examples are time based)? Thanks for any input. Garth ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Dialplan - strip IDD prefix and insert another
SellVoIP are great. Actually the rates are fine, but I like the quality as well. I can't say I ever needed phone support. - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, April 18, 2006 21:54 Subject: Re: [Asterisk-Users] Dialplan - strip IDD prefix and insert another How's tou're service with Sellvoip, I was not able to intergrate them into my system and they had no phone support. I'm using Gafachi now but prefer the rates Sellvoip provide. -- Original message -- From: "AR Tarzi" [EMAIL PROTECTED] SellVoIP appears to follow a US dialplan. A US numberis dialled as 1NXXNXX whereas an international (to the US) numberis dialled as 011X. Frankly, I didn't ask whether international numbers like Barbados where the code remains as 1 butare international (to the US) need the 011 or can be dialled directly but that's not really my concern. I've assumed they don't. Most of the world uses 00 as the internation prefix code, therefore I have to ask: Howcan I "strip" the 00 and insert 011 in one entry in the dialplan. I'm stripping the 00 and passing the rest of the numbers for numbersdialled as001X. (as in: 00|1XX.) but in case of numbers out of the US, how would I insert the 011 ? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] oh323 - unable to install
I'm and [EMAIL PROTECTED] user - been so now for almost a year. Lately, I've upgraded to the latest greatest.. (which is built on 1.2.5) and am unable to install oh323. I've already asked over at the ([EMAIL PROTECTED]) Sourceforge forum but no one seems to think it worth answering. The error I get is pretty obvious but I don't know where to go from here. More importantly, I need to have a workable solution - My question to the [EMAIL PROTECTED] gang was whether oh323 works (before I actually tried to install it). Here's the error (or where it starts). vpwlib/ChangeLog checking for g++... no checking for c++... no checking for gpp... no checking for aCC... no checking for CC... no checking for cxx... no checking for cc++... no checking for cl... no checking for FCC... no checking for KCC... no checking for RCC... no checking for xlC_r... no checking for xlC... no checking for C++ compiler default output... configure: error: C++ compiler cannot create executables See `config.log' for more details. make: *** No rule to make target `clean'. Stop. make: *** No rule to make target `opt'. Stop. checking for g++... no checking for c++... no checking for gpp... no checking for aCC... no checking for CC... no checking for cxx... no checking for cc++... no checking for cl... no checking for FCC... no checking for KCC... no checking for RCC... no checking for xlC_r... no checking for xlC... no checking for C++ compiler default output... configure: error: C++ compiler cannot create execu tables See `config.log' for more details. make: *** No rule to make target `clean'. Stop. make: *** No rule to make target `opt'. Stop. for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.5/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323V ERSION=\1.13.5\ -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/ope nh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o make[1]: g++: Command not found make[1]: *** [wrapper_misc.o] Error 127 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.5/wrapper' make: *** [subdirs_build] Error 1 for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.5/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323V ERSION=\1.13.5\ -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/ope nh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o make[1]: g++: Command not found make[1]: *** [wrapper_misc.o] Error 127 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.5/wrapper' make: *** [subdirs_build] Error 1 --- Installing GnuGK --- cp: cannot create regular file `/usr/sbin/gnugk': Text file busy mkdir: cannot create directory `/var/log/gk/': File exists STOPPING ASTERISK -- Remote UNIX connection Disconnected from Asterisk server Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started H.323 support installed. BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD
Re: [Asterisk-Users] Polycom IP 301 is slow
I've read all the messages here.. seems everyone's forgotten the original problem. Yes, the polycoms are very slow at rebooting when configuring through its http server, not only that, but it's also inconvenient because it reboots for every change (each page) which means you have to pass through several reboots to get it going. The overwhelming majority of people who use these phones (and swear by them) use an ftp server. This not only provides you with a record of the settings, but the ability to control a multitude of functions that are not reachable by browser. Examples: Volume of handset - if you don't do it via ftp, you have no way of setting it to store the volume between calls and the default volume I've found less than optimal Since you need to upgrade the phone(s) from time to time, and there's no firmware upload facility using a browser. To add ring tones (although I've not tried that one) These are only examples.. the ftp configuration allows you to have a default setup file for the site and a specific one for each phone so you don't really have to think of everything for the latter. I do not think the polycoms are the best at setting up but once you get them going, they sure are hard to beat. I personally like the speakerphone's quality (if you have an issue with it, yours MUST be faulty). - Original Message - From: Nick Hoffman [EMAIL PROTECTED] To: asterisk-users Mailing List asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 11:06 Subject: [Asterisk-Users] Polycom IP 301 is slow Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and find that it's extremely slow for configuring. For instance, it takes several minutes to boot up, apply any changes via the web interface takes at least a minute, etc. Is this normal behaviour? Is there anything that can be done about it? Thanks, -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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] GSM/DECT handsets (was gsm picocells)
Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. try www.imate.com (to start with) .. they have at least three types of GSM phones that do Wifi .. They run windows so there are several sip softwares and one IAX software that work with these - Also Nokia has a GSM phone that does Wifi but that's a symbian (OS) phone (don't know of sip software that works with it). - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 00:48 Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) If you find anything out, I would like to know. I have tried to find a gsm/wifi phone in the past (in melbourne) and failed. later, Paul Hales Technical Manager AsteriskIT - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 11:21 AM Subject: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the point being useless (or maybe I'm just in a bad mood today :) Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 24 March 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: gsm picocells Steve, Excellent explanation. In a nutshell, it might be better to just use a phone that can automatically switch between GSM and WiFi. Of course, that's limited to handful of handsets. I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? 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 ___ --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
Re: [Asterisk-Users] stop monitor on transfer
Trading desk environments are always recorded. This is for conflict resolution and there is no advice to clients. It is only used if the client claims are contrary to the trader's - therefore where a loss is concerned. Rather than test the legality, it is meant to resolve matters before they become a legal issue. The client, in some cases, is another institution with another call recorder, so it is also used to verify the traders' claims. Recording is a source of comfort. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 15:06 Subject: Re: [Asterisk-Users] stop monitor on transfer I'd teach the boss to appreciate recorded calls and just ensure they are secure. In the US I think this illegal? Aren't you supposed to have some sort of notification or beeping to indicate a recorded call to the other party? Not necessarily; there are some businesses that are required to record all conversations. One example are those involved with stock trading and the SEC regulations. Not sure what qualifies as notification. I'd suspect that appropriate wording in some privacy policy mailed to all clients might be sufficient, but that's a guess. There are a fair number of senior mgmt types that don't want to become another Enron case, and would rather not have any evidence of over-selling products, company stock, etc, for obvious reasons. Doctors even become nervous relative to recordings as a large percentage are only used for negative purposes. ___ --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] difference between records in CDR and realduration of call
That's because the duration is counted from the time of dialling. billsec is what you want if it's to calculate the duration the call was active. To change what shows you need to change call-log.php in /var/www/html/admin/cdr/ Instead of duration extract billsec - you can still label it duration (as in the title of the column). It might be closer to what you wish. - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 10, 2006 17:04 Subject: [Asterisk-Users] difference between records in CDR and realduration of call hi i've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? 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 ___ --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] Inserting access codes as prefixes to CID
When I receive a call from fwd, I'd like to insert a prefix prior to the caller ID - 1) to be able to look it up in a database ofidentified numbers and 2) for the receiver to be able to dial it back. So what I need is to identify the DID and based on that, insert the prefix. Any pointers to documentation would be appreciated BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTEzNTFaFw0wNjExMTEwOTEz NTFaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAeWFob28u Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQCvGOn8FwM/UUm7OYMdFZYn+hUrmDYo ARJGJvFDu7lnbrT/v3tf1zRpOULT8yN2PXtSUmsxlvYX2SCJ8PggECGGbyJEkd8bHmPJEi7g FHNs9h3ps7SJ+gQFkqa0soxegfHgQzrjrOGXNI1dMCKaYc6a2dSWRUBj4C1ii1dHYs7jmQID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQHlhaG9vLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAC9Tm59BZjKmw61xcYa4yXhPSqfkXTJy6eAVX4LSwM1gkRbV6HWZ HjQBmEhTkfrAF01xeKrDRh6vJIYGjSuPJRVmCN2+BA/UuNnK3EQOI+mwuku8KQzDAFXpJHhe +J5626T7NiuADtT2F0L3tLoFf8vvLcyTzvCHU+y6E2Danaak KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZXMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTE5MDRaFw0wNjExMTEwOTE5 MDRaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAZ21haWwu Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDASKRiH2YqhCqPF3HDlPCdtHZb78Pn Z4S/qzgdLVdzeE1b2Ddd4gl+FkQw2IS4Q+3XSwsGyh9wY6irNb+nIrr5Gs9+JmpQTSPjQp72 trLvD+PvFetwQMotRODVsgxHIpgcTFBjpMZ4P24NeAGRBNzfPjwqx3gfscd10fWtiXGo8wID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQGdtYWlsLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAAZ2rAEswRkNEgiMcy3enKlTcQ9QiIFeQP5bq7iXDUkbhtcZHDdi ol+HaN6QyO2ZUCYbuK1d12VD92QpZuRxw0lS7K7qWU7aF5gabpnEjl1KQ0ujr+gEcV2ogvZY 2F4SZ7H9uF0c06/NT5TpoFyok3wJ/jZXJhRAbR/Eye678OCq KEY;X509;ENCODING=BASE64: MIICfDCCAeWgAwIBAgIDD80vMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMDYyMTMzMzVaFw0wNjExMDYyMTMz MzVaMG8xDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxJTAjBgkqhkiG9w0BCQEWFmFydGFyemlAYmF0ZWxj by5jb20uYmgwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAK+koXkgs50JRrsTV4tj2QS7 uZ05+iKe/lhkdv56a6oEUcw4tO03rGMcB+ocWwfmmIbZ1n5p8dRjybsZMI5zEnRsf/KeQLl3 1wBPYoKzVDQrulNMGh8FmhK8uWsW1FZSKJkbxZWjcI2fkbDLmQuvWBUdlgiOFOLp08m9bMvf ZpCfAgMBAAGjMzAxMCEGA1UdEQQaMBiBFmFydGFyemlAYmF0ZWxjby5jb20uYmgwDAYDVR0T AQH/BAIwADANBgkqhkiG9w0BAQQFAAOBgQA/TNRreOLNx7d1f7H9vfrnlTRuftVHVL4f6h6X u2Od18TDDP6/iUuiTtcMQfOOwiBBxjkgdupsDi4q8FrOseWu5ylM9hNg+1mtjSQT00CL6n4A CIh94LiywiMeJmxzKLuihUxyQu2aRFksaQS4unmENCZ23a+xB4DHuTD9V3FcAw== EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;PREF;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] REV:20060305T155747Z END:VCARD ___ --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] Dialplan - strip IDD prefix and insert another
SellVoIP appears to follow a US dialplan. A US numberis dialled as 1NXXNXX whereas an international (to the US) numberis dialled as 011X. Frankly, I didn't ask whether international numbers like Barbados where the code remains as 1 butare international (to the US) need the 011 or can be dialled directly but that's not really my concern. I've assumed they don't. Most of the world uses 00 as the internation prefix code, therefore I have to ask: Howcan I "strip" the 00 and insert 011 in one entry in the dialplan. I'm stripping the 00 and passing the rest of the numbers for numbersdialled as001X. (as in: 00|1XX.) but in case of numbers out of the US, how would I insert the 011 ? BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTEzNTFaFw0wNjExMTEwOTEz NTFaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAeWFob28u Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQCvGOn8FwM/UUm7OYMdFZYn+hUrmDYo ARJGJvFDu7lnbrT/v3tf1zRpOULT8yN2PXtSUmsxlvYX2SCJ8PggECGGbyJEkd8bHmPJEi7g FHNs9h3ps7SJ+gQFkqa0soxegfHgQzrjrOGXNI1dMCKaYc6a2dSWRUBj4C1ii1dHYs7jmQID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQHlhaG9vLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAC9Tm59BZjKmw61xcYa4yXhPSqfkXTJy6eAVX4LSwM1gkRbV6HWZ HjQBmEhTkfrAF01xeKrDRh6vJIYGjSuPJRVmCN2+BA/UuNnK3EQOI+mwuku8KQzDAFXpJHhe +J5626T7NiuADtT2F0L3tLoFf8vvLcyTzvCHU+y6E2Danaak KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZXMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTE5MDRaFw0wNjExMTEwOTE5 MDRaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAZ21haWwu Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDASKRiH2YqhCqPF3HDlPCdtHZb78Pn Z4S/qzgdLVdzeE1b2Ddd4gl+FkQw2IS4Q+3XSwsGyh9wY6irNb+nIrr5Gs9+JmpQTSPjQp72 trLvD+PvFetwQMotRODVsgxHIpgcTFBjpMZ4P24NeAGRBNzfPjwqx3gfscd10fWtiXGo8wID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQGdtYWlsLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAAZ2rAEswRkNEgiMcy3enKlTcQ9QiIFeQP5bq7iXDUkbhtcZHDdi ol+HaN6QyO2ZUCYbuK1d12VD92QpZuRxw0lS7K7qWU7aF5gabpnEjl1KQ0ujr+gEcV2ogvZY 2F4SZ7H9uF0c06/NT5TpoFyok3wJ/jZXJhRAbR/Eye678OCq KEY;X509;ENCODING=BASE64: MIICfDCCAeWgAwIBAgIDD80vMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMDYyMTMzMzVaFw0wNjExMDYyMTMz MzVaMG8xDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxJTAjBgkqhkiG9w0BCQEWFmFydGFyemlAYmF0ZWxj by5jb20uYmgwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAK+koXkgs50JRrsTV4tj2QS7 uZ05+iKe/lhkdv56a6oEUcw4tO03rGMcB+ocWwfmmIbZ1n5p8dRjybsZMI5zEnRsf/KeQLl3 1wBPYoKzVDQrulNMGh8FmhK8uWsW1FZSKJkbxZWjcI2fkbDLmQuvWBUdlgiOFOLp08m9bMvf ZpCfAgMBAAGjMzAxMCEGA1UdEQQaMBiBFmFydGFyemlAYmF0ZWxjby5jb20uYmgwDAYDVR0T AQH/BAIwADANBgkqhkiG9w0BAQQFAAOBgQA/TNRreOLNx7d1f7H9vfrnlTRuftVHVL4f6h6X u2Od18TDDP6/iUuiTtcMQfOOwiBBxjkgdupsDi4q8FrOseWu5ylM9hNg+1mtjSQT00CL6n4A CIh94LiywiMeJmxzKLuihUxyQu2aRFksaQS4unmENCZ23a+xB4DHuTD9V3FcAy== EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;PREF;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] REV:20060305T155750Z END:VCARD ___ --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] Dialplan - strip IDD prefix and insert another
Thank you. works like a charm. I'm using [EMAIL PROTECTED] so I had to massage
it into AMP's structure.
Your example is actually the reverse of what I needed to do, but that's not
the issue.
AMP uses a macro to dial (syntax almost exactly the same).
I feel this should be documented somewhere (been googling all day) - so much
appreciated.
- Original Message -
From: Ira [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, March 05, 2006 21:01
Subject: Re: [Asterisk-Users] Dialplan - strip IDD prefix and insert another
At 07:57 AM 03/05/2006, you wrote:
How can I strip the 00 and insert 011 in one entry in the dialplan. I'm
stripping the 00 and passing the rest of the numbers for numbers dialled
as 001X. (as in: 00|1XX.) but in case of numbers out of the US, how would
I insert the 011 ?
exten = _011X. , 1, dial(sip/1/00${EXTEN:3})
Or something similar to that. Match to the 011, delete it, {EXTEN:3}, and
then add the 00 before dialing.
Ira
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[Asterisk-Users] Polycom bootrom and SIP software
I know this shouldn't be the place to ask this, but I've just tried to upgrade my IP600 with bootrom 2.6.2 and SIP 1.5.2 and I'm getting intotrouble here (I chose not to go to the higher software levels since there's a warningabout using"secure" links.. I am not trying to change anything functionally but thishas been a long outstanding upgrade. Now when I did the upgrade I found no BootRom.ver The log files(one of them anyway) seem to indicate theproblemis with loading the BootRom.ld I'm a complete layman with this. I do not know what the bootrom.ld does, what the others do..but the phone is in a loop now (gives an error and tries to reboot). If this is somehow a known problemhit when upgrading, I should have seen it from the long history of messages I have. Any help would be appreciated. I could post things but I think it better to find someone who knows what they want to read first. BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTEzNTFaFw0wNjExMTEwOTEz NTFaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAeWFob28u Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQCvGOn8FwM/UUm7OYMdFZYn+hUrmDYo ARJGJvFDu7lnbrT/v3tf1zRpOULT8yN2PXtSUmsxlvYX2SCJ8PggECGGbyJEkd8bHmPJEi7g FHNs9h3ps7SJ+gQFkqa0soxegfHgQzrjrOGXNI1dMCKaYc6a2dSWRUBj4C1ii1dHYs7jmQID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQHlhaG9vLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAC9Tm59BZjKmw61xcYa4yXhPSqfkXTJy6eAVX4LSwM1gkRbV6HWZ HjQBmEhTkfrAF01xeKrDRh6vJIYGjSuPJRVmCN2+BA/UuNnK3EQOI+mwuku8KQzDAFXpJHhe +J5626T7NiuADtT2F0L3tLoFf8vvLcyTzvCHU+y6E2Danaak KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZXMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTE5MDRaFw0wNjExMTEwOTE5 MDRaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAZ21haWwu Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDASKRiH2YqhCqPF3HDlPCdtHZb78Pn Z4S/qzgdLVdzeE1b2Ddd4gl+FkQw2IS4Q+3XSwsGyh9wY6irNb+nIrr5Gs9+JmpQTSPjQp72 trLvD+PvFetwQMotRODVsgxHIpgcTFBjpMZ4P24NeAGRBNzfPjwqx3gfscd10fWtiXGo8wID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQGdtYWlsLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAAZ2rAEswRkNEgiMcy3enKlTcQ9QiIFeQP5bq7iXDUkbhtcZHDdi ol+HaN6QyO2ZUCYbuK1d12VD92QpZuRxw0lS7K7qWU7aF5gabpnEjl1KQ0ujr+gEcV2ogvZY 2F4SZ7H9uF0c06/NT5TpoFyok3wJ/jZXJhRAbR/Eye678OCq KEY;X509;ENCODING=BASE64: MIICfDCCAeWgAwIBAgIDD80vMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMDYyMTMzMzVaFw0wNjExMDYyMTMz MzVaMG8xDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxJTAjBgkqhkiG9w0BCQEWFmFydGFyemlAYmF0ZWxj by5jb20uYmgwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAK+koXkgs50JRrsTV4tj2QS7 uZ05+iKe/lhkdv56a6oEUcw4tO03rGMcB+ocWwfmmIbZ1n5p8dRjybsZMI5zEnRsf/KeQLl3 1wBPYoKzVDQrulNMGh8FmhK8uWsW1FZSKJkbxZWjcI2fkbDLmQuvWBUdlgiOFOLp08m9bMvf ZpCfAgMBAAGjMzAxMCEGA1UdEQQaMBiBFmFydGFyemlAYmF0ZWxjby5jb20uYmgwDAYDVR0T AQH/BAIwADANBgkqhkiG9w0BAQQFAAOBgQA/TNRreOLNx7d1f7H9vfrnlTRuftVHVL4f6h6X u2Od18TDDP6/iUuiTtcMQfOOwiBBxjkgdupsDi4q8FrOseWu5ylM9hNg+1mtjSQT00CL6n4A CIh94LiywiMeJmxzKLuihUxyQu2aRFksaQS4unmENCZ23a+xB4DHuTD9V3FcAx== EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;PREF;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] REV:20060227T131602Z END:VCARD ___ --Bandwidth
Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 111
yikes, 25 and 110 will allow mail - but please without the whole digest attached. And wouldn't your question be more useful with a better subject field? now no one will see me addressing a question I know an answer to. ___ --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 - Skype anywhere/anyhow?
could you please tell how it interfaces with Asterisk? Could I receive calls into Asterisk? send calls out? I've just downloaded it and am searching (unsuccessfully) for these on Gizmo's site/software. - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, December 19, 2005 23:23 Subject: RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? Everyone should simply uninstall Skype and switch to the Gizmo project because it interfaces quite nicely with Asterisk. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.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] Can Asterisk replace Cisco 5350?
Ignoring SS7, why exactly are you setting up several boxes ? there are quad E1 cards no ? This is way out of my league, but I just want to understand. - Original Message - From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 17, 2005 12:19 Subject: Re: [Asterisk-Users] Can Asterisk replace Cisco 5350? Linuxnizer The Mesmorizer a écrit : Hi, We are using Cisco5350 as a gateway with 2 E1 cards (part# AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my question is can we save some money and use Asterisk + PCI E1 cards? I've had the same issue lately. I need to set up a 4E1 / g.729 solution. Asterisk way - 4 asterisk boxes with 1 E1 card (approx $2k each) - 120 g.729 licences ($1.2k) - 1 SER box to dispatch the calls on the 4 asterisk boxes ($1k) Total: 4 * 2 + 1.2 + 1 = 10.2 In the end, I went on voipsupply.com and saw that they offer Audiocodes mediant 2000 (4 E1 version) for about $10k. Since this box also does SS7 (which is nice if you want to properly interconnect some day), can scale up to 16 E1 and is conveniently packed in a 1U rackable unit, I have decided to go with Audiocodes. Since I am not set up yet, I can't tell wether it is a good decision or not. I will let you know :) Cheers, Jean-Michel. ___ --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] DISA function
I had a problem with DTMF with DISA.. I am using a Sipura SPA 3000 for the line. I set the FXO port impedance (on the PSTN line tab) to 900 as advised by others and it worked. Having said that, I'm sure you will be using some other FXO adapter.. Just thought I'd tell. - Original Message - From: Richard Smith To: asterisk-users@lists.digium.com Sent: Monday, December 05, 2005 01:44 Subject: [Asterisk-Users] DISA function Hi all, I was wondering whether the DISA function on the latest asterisk 1.2 stable release actually works better than the other prior releases. Basically the [EMAIL PROTECTED] version 2.0 BETA 4 I'm using does not recognise the DTMF tones all the time and sometime when it does, it disconnects. Cheers, Richard. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Polycom DTMF after connection not working
On a polycom 600 which is working perfectly otherwise, I am unable to use DTMF with IVR or such - not even to dialout of a Sipura setup elsewhere. Other phones (analogue connected to ATA) are accepted. I suspectthe phone is not using rfc2833 but I don't know how to specify that it should useit (not available on the http configuration). ___ --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] Incoming call trunk fwd not work
By observation (or better said, I don't know why) clue:719705 @fromiaxfwd ?? Your peer and user settings should be taken from www.freeworlddialup.comwhich dictate that your user context should be [iaxfwd] NOT your userID (or fwd number) They also have settings related to authentication and such.. I assumed that you had done that. - Original Message - From: Cristian Paun To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, November 17, 2005 03:07 Subject: RE: [Asterisk-Users] Incoming call trunk fwd not work I did that but still same error message Nov 16 19:06:14 NOTICE[1326]: Rejected connect attempt from 192.246.69.187, request '[EMAIL PROTECTED]' does not exist Can I do something else? Cristian Paun, MCSE K2 systems inc. Tel: 514 745-6000 Fax: 514 745-9000 Cell: 514 963-5279 [EMAIL PROTECTED] www.k2systems.ca From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AR TarziSent: November 15, 2005 8:35 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Incoming call trunk fwd not work In AAH create a DID using the number 719705 and direct it to ring wherever you wish it to (extension.. extension group etc.) - Original Message - From: Cristian Paun To: asterisk-users@lists.digium.com Sent: Tuesday, November 15, 2005 21:08 Subject: [Asterisk-Users] Incoming call trunk fwd not work I have an AAH installed with trunk FWD. I am able to place calls but not receive. I get these message Nov 15 13:05:52 NOTICE[1410]: Rejected connect attempt from 192.246.69.187, request '[EMAIL PROTECTED]' does not exist My AAH box is in NAT mode Can somebody give me a config file worked with FWD account in NAT mode? Thanks in advance Cristian ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming call trunk fwd not work
In AAH create a DID using the number 719705 and direct it to ring wherever you wish it to (extension.. extension group etc.) - Original Message - From: Cristian Paun To: asterisk-users@lists.digium.com Sent: Tuesday, November 15, 2005 21:08 Subject: [Asterisk-Users] Incoming call trunk fwd not work I have an AAH installed with trunk FWD. I am able to place calls but not receive. I get these message Nov 15 13:05:52 NOTICE[1410]: Rejected connect attempt from 192.246.69.187, request '[EMAIL PROTECTED]' does not exist My AAH box is in NAT mode Can somebody give me a config file worked with FWD account in NAT mode? Thanks in advance Cristian ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check how many G729 codec license installed
- Original Message - From: Angelito Manansala [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, November 13, 2005 14:31 Subject: [Asterisk-Users] How to check how many G729 codec license installed Guys, is the any CLI commands or info files where you can check how many g729 codec license installed. Regards, Lito show g729 If you've bought them from Digium, they'd have provided these instructions. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: * Mobile Phone Mobile Network
I've used a Nokia 32 unattended (remote) for the past year or so. David Uzzell [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] | Ok I have a question. Seen it come and go around the mailling list for a | while but never really seen an answer that seems to sort it out. | | What is needed is some interface from * Mobile Phone Mobile Network | Service. | | At this point all the providers in AUS that I have found are charging a | Premium Rate for Land Line Mobile Network services. | | What I would like to do is be able to purchase a low rate Mobile SIM | that I can chuck into a Mobile Phone and have it setup so that I route | the Mobile calls through it. | | Rembering that most if not all mobile phones can be accessed via RS232 | interface. | | Anyone done this or seen it done or know how to do it using * and whatever? | | Cheers | David | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Mobile Callings
Nokia's 32 is just one example. A bit pricey but reliable. - Original Message - From: Germán Micale Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Monday, January 24, 2005 22:20 Subject: RE: Mobile Callings Thank you Andrew,The network is gsm-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] En nombre de AndrewThompsonEnviado el: lunes, 24 de enero de 2005 17:24Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] Mobile CallingsGermán Micale wrote: Hi, Does someone knows what kind of device I need to call from my pc to the mobile network? In Spain VoIP prices are very similar to call to a mobile than do it from an other mobile. So, I want to plug some device to the PC and get out the call throught it, but I dn't know what kind of device I need. Thanks in advancelook up: cellsocketThere are other similar devices, but the names now slip my mind. What type of network is your cell phone on? (cdma, gsm, tdma, etc)-- Andrew Thompsonhttp://aktzero.com/http://dev.asteriskdocs.org/___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM to ISDN or TAPI
I use Nokia 32s. I don't know what a fritz card is but they can act either as an FXO or as an FXS device. Beware though, if you use them off a port that expects a telephone set (like an ATA or so) you'll need a special cable to program the 32 properly - the cable is pricey.Acting asanFXO, they perform out of the box (don't need the cable). - Original Message - From: Chris Lee To: [EMAIL PROTECTED] Sent: Thursday, June 10, 2004 17:49 Subject: [Asterisk-Users] GSM to ISDN or TAPI HiI am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface.I will even plug two devices into a windows box and have that do ISDN to ISDN if required.All I want is two GSM lines that look like voice modems to the PC and provide full telephony interface, that is DTMF both ways CLI and a few other bits and pieces.I am looking to using asterisk as a remote IVR for looking after some equipment, but land lines are a problem.Any help is much appreciatedRegardsChris.___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: Re: [Asterisk-Users] X100P Ireland Red Alarm
Important. 1. Try the phone (set) directly on the line.. - confirm you have dialtone 2. Make sure the phone is picking up the line from pins 3 4 on the RJ11 ONLY .. i.e. if your line is using a non-standard interface (and so does your phone) this is a possible failure - not of the card, but of the connection. I must emphasize, I have absolutely no experience with *, Digium hardware, and very little experience with the general subject. I do, however, have some experience trying to connect phones to lines.. (outside the US) - Original Message - From: Eric Wieling To: [EMAIL PROTECTED] Sent: Saturday, May 15, 2004 22:23 Subject: Re: Subject: Re: [Asterisk-Users] X100P Ireland Red Alarm On Sat, 2004-05-15 at 14:01, Aaron Clauson wrote: I suspected that I the analogue phone should have got a pass through signal when the power was off to the server, unfortunately it doesn't. I kept asking digium support about that but they didn't give me an answer. The problem is how do I identify whether the X100P is incompatibel with the network or faulty without possibly wasting another USD100???The two ports on the X100P are *HARDWIRED TOGATHER*. If plugging aregular analog phone into the second port does not give you a dialtonethen either some of the traces on the board are broken or the line isnot working. I assume you've confirmed the line if fine by plugging ananalog phone directly into the line?-- Eric Wieling * BTEL Consulting * 504-899-1387 x2111"In a related story, the IRS has recently ruled that the cost of Windowsupgrades can NOT be deducted as a gambling loss."___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Ireland Red Alarm (AR Tarzi)
Note: some lines provide "ring" on a third wire. If that's the case, need to bridge that wire to 3 or 4 (the middle pins in 123456). I'm sorry I can't be more specific (or even describe why).. I'm just alay personwho's interested but now curious .. Bumble wha ?? - Original Message - From: Aaron Clauson To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 00:17 Subject: Re: [Asterisk-Users] X100P Ireland Red Alarm (AR Tarzi) Ahhh this could be my problem! I just checked whichwires on the RJ11 cable had a voltage across them andit was the yellow and green (3 4?). From whatsomeone posted the other day it's supposed to beBumble Bee and Christmas Tree.I did have to get a technician out to fix my line whenit was first installed because it was dead. Maybe hewired it up incorrectly or maybe they just do itdifferent here in Ireland.I'll buy a crimping tool tomorrow and try outdifferent combinations.Thx.Aaron From: "AR Tarzi" [EMAIL PROTECTED]Important.1. Try the phone (set) directly on the line.. -confirm you have =dialtone2. Make sure the phone is picking up the line frompins 3 4 on the =RJ11 ONLY .. i.e. if your line is using anon-standard interface (and so =does your phone) this is a possible failure - not ofthe card, but of =the connection.__Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.http://promo.yahoo.com/sbc/___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A General question
Could someone find the time to tell me whether ALL functions in Asterisk are programmed using scripts and contexts ? What I need to find out is whether the userCAN configure services for themselves.. Below, I chose a sampling of a fresh question and answer (just as an example). In this example: Is it not possible for each user to configure (say) the voicemail so that whenever there's a message (data which probably affects the dialtone or a light on the recipient phone), an action is taken.. A daemon could then check a database to find out the action (dialout, sms message.. or whatever) and the target number. I cannot imagine the size of user installations (observed in user messages) being managed centrally.. I hope this is not too much to ask, but all I need is a cryptic yes or no at this stage. - Original Message - From: Andrew Kohlsmith To: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 17:33 Subject: Re: [Asterisk-Users] Dialing out after caller leaves message I cannot use Dial after the Voicemail application, e.g., [Step 1] exten = 100, 1, Dial( SIP/100, 15 ) [Step 2] exten = 100, 2, Voicemail( u100 ) [Step 3] exten = 100, 3, Dial( Zap/g1/CELL_PHONE ) because the caller will hang up after leaving the voice mail in Step 2 above, and Asterisk will terminate the script, so Step 3 will get never get executed.What about putting this in a special context and using 'h'?i.e.exten = 100, 1, Dial(SIP/100, 15)exten = 100, 2, Voicemail(u100)exten = h,1,Dial(Zap/g1/CELL_PHONE)? h will get executed on hangup. The only caveat is that if no voicemail was left, you will still get called. Is there some way to check if there are new messages and use that in h along with the Dial()?Regards,Andrew___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
