[asterisk-users] sangomo
Hi everyone I just ordered a Sangoma A20001 with 2FXO ports - Does anyone have suggested reading pointers for what I'll need to do to get it working? I've only used VoIP in the past so don't know much about Sangoma drivers or Zaptel. I opted for the non-echo canceling card so I may need to do some tuning? Looking for reading... Hurl an URL at me! thanks! Todd ___ --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] BLF on GXP2000
While I don't see anything wrong with this, I'm no expert. I took my instructions from the following URL and they worked fine... I have the subscribecontext in General and it works fine. What is the firmware on the GXP? old firmware may be related -t- http://www.jackenhack.com/blog/archives/2005/11/22/setting-up- subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/ ___ --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] (no subject)
Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd ___ --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] Question about hardware
The card will let you interface with a regular telephone line instead of VoIP. If you want to use a regular phone instead of the computer softphones, look into the Grandstream handytone devices - they'll make it so your regular telephones can talk to Asterisk. You can make the system work fine with softphones so there's no additional cost at this point... Todd I ordered the card off ebay. Is there anything else I'd need - special cords, phones, etc? I'd have to try for them next month or after, but I'd prefer to know what they are now so that I can be looking for 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] Question about hardware
Speaking of the X100P, I am going to setup an asterisk server next week for a friend's business to replace his aging system. He currently has two voice lines and another line for the fax machine. I was looking at the Sangoma A20200D but that's pretty expensive... We're going to use Grandstream GXP's on desks... Do I need hardware echo cancelation (I'm thinking of using a Dell 2.0 GHz machine)? As Asterisk can handle fax, I was going to drop the 2nd voice line, have the phone company roll busy onto the current fax line, and use that as the second voice line. Can I just use two of the X100 cards? Or is that asking for trouble? thanks Todd On Dec 13, 2006, at 9:56 AM, John Novack wrote: Don't forget that IF you have NO card, you need to roll ZTDUMMY into the compile. With no card though, you will not be able to read the incoming CLID Also, IF you ever want to progress beyond the X100 card, The Digium cards ( beyond your present budget ( are really intolerant of older PCI buses. Sangoma works with MANY more motherboards. John Novack ___ --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 help getting started with asterisk
As Paul Hales said, I doubt that modem is supported. To interface with a regular phone line you'll need to get a supported card. You can read about it online. To just get started with playing, I recommend you go ahead with the sophistocated VoIP stuff.. Perhaps sign up with IPKALL or Stanaphone. Google will tell you all about how to connect them... Good luck and have fun! Todd On Dec 12, 2006, at 10:13 PM, Michael Sullivan wrote: I am new to asterisk. I need help getting started, if it's even worth getting started. I say if it's worth getting started because I'm not sure if my hardware will even work with asterisk. I have a US ROBOTICS 56K V.90 PCI SOFT MODEM. I have standard twisted pair telephone wire. I can't afford to alter my hardware. I know I won't be able to do any sophistocated VoIP stuff. All I want is for asterisk to provide caller ID information for my Gentoo box and to drop calls for certain phone numbers I specify. Someone on the Gentoo list told me that for the caller ID bit I should check my modem's manual to find out how; my modem did not come with a manual. I've emerged asterisk on my Gentoo system, as well as the zaptel driver package. I issued /etc/init.d/asterisk start and confirmed with ps that it was running. I got a command console with asterisk -r. I then called my home line that the computer is plugged into from my cell phone. I heard my wife's cordless ringing in the other room, and I heard some breaks on the phone line, but nothing else. I let it ring ten times. What would I have to do to get asterisk to realize that the PC is connected to the phone line? I know it is because I can dial out with kppp. Also, I've been trying to follow the AsteriskTFOT.pdf file. On page 79 it says to add a few lines to /etc/zaptel.conf and then modprobe wctdm. I did that. I then ran /sbin/ztcnf -vv to make sure everything was right. I got this: camille ~ # modprobe wctdm camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 1 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) What does that mean? What device was it looking for? Please help! -Michael Sullivan- ___ --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] CLI History
short version: me too long version: The same thing happens on my asterisk boxes - both built with the latest trixbox image... perhaps that's a factor? My history is always restart now, although I typically connect and run sip show peers. I haven't typed restart now in a long time, but that is the first thing when I hit up-arrrow upon connecting I have had history written to when I type 'exit' at the console instead of ctrl-c. I haven't tested though as the school bus just arrived ;) Todd On Dec 11, 2006, at 3:35 PM, Douglas Garstang wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Monday, December 11, 2006 12:57 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CLI History On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote: On Monday 11 December 2006 9:31 am, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI A No such command 'A' (type 'help' for help) hera*CLI B No such command 'B' (type 'help' for help) hera*CLI C No such command 'C' (type 'help' for help) hera*CLI D No such command 'D' (type 'help' for help) hera*CLI E No such command 'E' (type 'help' for help) hera*CLI [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. == === Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI stop now -- I pressed the UP arrow upon re-entering the console! I'm a bit confused by your example. What are A,B,C, etc? To exit the Asterisk console, I type 'exit'. Asterisk continues to run, as it should. To re-enter the console I use asterisk -rvvv. He was demonstrating how the CLI history shows stop now as the last command (which um... it's a history? you're last command is gonna be the um... last command you ran... i.e. stop now). For crying out loud, why is this so hard to understand? It isn't rocket science. I said that when I exit the CLI and re-enter, no matter what my previous set of commands was, when I hit the UP arrow key, it was always 'stop now'. 'Stop now' WAS NOT MY PREVIOUS COMMAND. For the person that suggested maybe unknown commands are not added to the history... hera*CLI show channels Channel Location State Application(Data) 0 active channels 0 active calls hera*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 0 active SIP channels hera*CLI (I Pressed Ctrl-c here) [13:[EMAIL PROTECTED](pbx3):~]# asterisk -trv Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. == === Connected to Asterisk 1.2.9.1 currently running on hera (pid = 18149) Verbosity is at least 3 hera*CLI stop now (I pressed the up arrow key here) As you can see, my previous commands where 'show channels' and 'sip show channels'. When I exited the CLI and re-entered and pressed ctrl-c, the commands in the history where not 'show channels and 'sip show channels' but 'stop now' instead. 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
Re: [asterisk-users] Dual Wan Router with Failover
I've been looking for this as well.. I need to support up to 20 VOIP phones over Internet as the Asterisk server is off-site. We'll have multiple cable modems or DSL routers. I found this device which looks promising - does anyone have any experience with this? http://www.peplink.com/productsLoader.php?productName=balance ToddOn Nov 13, 2006, at 8:49 PM, Dovid B wrote: Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid___ --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] config template for Grandstreams
Thanks- they did respond. I got a new template, but was asked to not share it for now - it'll be on their website in a few days pending committee approval thanks Todd On Nov 14, 2006, at 12:50 PM, Gordon Henderson wrote: On Fri, 10 Nov 2006, Todd- Asterisk wrote: I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate the config file from a template, but the templates posted on their website are for an old version of the phone firmware. Anyone have a tool or access to templates for the latest firmware versions? Email their technical support. I did this a few days ago for the latest one for the GPX2000 and they emailled it back the next day. 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 ___ --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] config template for Grandstreams
I see the Grandstream website now has the new config templates posted with all the happy P commands... http://grandstream.com/y-configurationtool.htm Todd ___ --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: Grandstream TFTP system wide settings
Perhaps you want to look at http://tanesha.net/Wiki/GratissipTftpd.html You can keep the P-codes in a mysql database and build all the configs you want. For me, it was a little too much work for the few phones I have, but if you need more. Todd Is there a quicker way to change settings for all Grandstream phones, is there any one file which can act as a global configuration file without changing each phones phone specific settings? And can't it be simply done by text editing, without the need to convert each file to cfgmac format? ___ --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] config template for Grandstreams
I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate the config file from a template, but the templates posted on their website are for an old version of the phone firmware. Anyone have a tool or access to templates for the latest firmware versions? I guess the procedure is to modify the template, then run the configuration tool on the template to generate the specific downloadable file..? Thanks Todd ___ --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] several behind NAT
Just to report back in, the advice of the list was to not worry about it- they should work well. I took a DSL modem with a router on it and connected both phones (Grandstream GXP2k and 101)- they did not work. I found that I had to program in a STUN server. I also has to set it to use a random port instead of the default- a pre-defined port (else only 1 phone would ring regardless of extension). Now they both work well. Does anyone see a problem with this setup? Should I use my own STUN server? or can I continue with stun.fwdnet.net? Also, where can I get information on provisioning? These phones will be out of my hands soon and I'd like to be able to update the configs. I saw a few utilities for generating the configs, but I'd like more specific info - I don't mind editing files by hand but want to know how it works. Does anyone have some resources? thanks for all the help- this is a great list. ToddOn Nov 6, 2006, at 10:28 AM, Todd- Asterisk wrote:I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server. Should I look into using STUN servers? Will this setup be a problem? I've read about NAT and STUN on voip-info but am looking for more information.. btw- I'm not set on Grandstream. If you think Polycom or something can handle NAT better, then I'll use that instead. I guess there's no IAX phones yet... Thanks in advance. Todd___ --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] several behind NAT
I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server. Should I look into using STUN servers? Will this setup be a problem? I've read about NAT and STUN on voip-info but am looking for more information.. btw- I'm not set on Grandstream. If you think Polycom or something can handle NAT better, then I'll use that instead. I guess there's no IAX phones yet... Thanks in advance. Todd ___ --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] Which IP phones have best voice quality, preferably under $150
I have the Budgetone 101 and GXP2000 and thought the sound quality was excellent. Even over the internet... I agree with Joe that something else may be the factor... Todd Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] getting DID info..
This might be a newbie question... I'm using a SIP trunk and trying
to get DID line information on an incoming call. All I hear is a
nice lady saying 'Zero' - then the call continues... Any suggestions?
thanks
Todd
exten = s,n,Set(DIDID=(${FROM_DID}))
exten = s,n,SayNumber(DIDID)
or
exten = s,n,Set(FROM_DID=${EXTEN})
exten = s,n,SayNumber(FROM_DID)
and a third try.. (I'm not sure what 's' is, but saw it somewhere..)
exten = s,n,Set(FROM_DID=s)
exten = s,n,Wait(1)
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Re: [asterisk-users] getting DID info..
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the
GoTo command is setting the extension to 's' when it transfers
control to the context defined in the IAX.conf -where I have the
trunk line defined...
exten = h,1,Hangup
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,SayAlpha(${EXTEN})
It is my impression that the EXTEN variable is used as the internal
extension - not the incoming DID number, but you seem pretty
confident so I must be wrong. What Im looking to do is a FOP pop-up
with the DID number and caller ID number in it... I'll tie that
into a web-based database...
Here's my full log file..
Oct 20 14:23:42 VERBOSE[5387] logger.c: -- Accepting
AUTHENTICATED call from 204.11.194.34:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw|gsm),
priority = mine
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Set
(IAX2/204.11.194.34:4569-4, LOOPCOUNT=0) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Set
(IAX2/204.11.194.34:4569-4, __DIR-CONTEXT=default) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Answer
(IAX2/204.11.194.34:4569-4, ) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Wait
(IAX2/204.11.194.34:4569-4, 1) in new stack
Oct 20 14:23:43 DEBUG[5387] chan_iax2.c: Ooh, voice format changed to 4
Oct 20 14:23:43 VERBOSE[5862] logger.c: -- Executing SayAlpha
(IAX2/204.11.194.34:4569-4, s) in new stack
Oct 20 14:23:43 DEBUG[5862] channel.c: Scheduling timer at 160 sample
intervals
Oct 20 14:23:43 VERBOSE[5862] logger.c: -- Playing 'letters/
s' (language 'en')
DID is the inbound call number.
The is notation for CallerID name, that won't help.
s is the start extension. setting it to FROM_DID makes no sense.
(This is the extention that starts in this context; it is a
default, if
the context is started without an extension. (eg batphone or called
from another
context))
FROM_DID=${EXTEN} gets you the right number.
However, SayNumber is looking for a SINGLE digit. Your
000-000- style number is overflow, and hence zero.
You have to parse the number to do this right.
If you aren't sure how, let me know, I might have a macro to do it.
Thanks,
J.
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Re: [asterisk-users] getting DID info..
Hi Eric- It wasn't typo, it was "truncated for posting" :) Below are the complete relevant files. I'm getting 'S' when I want to hear the DID number.. This machine was a trixbox about a two weeks ago, but I've since tossed away the GUI and do everything by hand now. I just used the trixbox for learning how this stuff works. I want to have the DID in some variable so I can pass it to FOP for a popup and link to our web-based database... I've used the free DID number from didww.com using IAX and SIP trunks. If 's' is the correct extension, as I expect it is, how do I get the DID number that the call came in on? ToddLog File during a call from an outside IAX line (form didww.com)Oct 20 15:32:44 VERBOSE[5387] logger.c: -- Accepting AUTHENTICATED call from 204.11.194.34: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mineOct 20 15:32:44 VERBOSE[9086] logger.c: -- Executing Set("IAX2/204.11.194.34:4569-4", "LOOPCOUNT=0") in new stackOct 20 15:32:44 VERBOSE[9086] logger.c: -- Executing NoOp("IAX2/204.11.194.34:4569-4", "8603492460") in new stack Oct 20 15:32:44 VERBOSE[9086] logger.c: -- Executing NoOp("IAX2/204.11.194.34:4569-4", "s") in new stackOct 20 15:32:44 VERBOSE[9086] logger.c: -- Executing Wait("IAX2/204.11.194.34:4569-4", "1") in new stackOct 20 15:32:45 VERBOSE[9086] logger.c: -- Executing SayAlpha("IAX2/204.11.194.34:4569-4", "s") in new stackOct 20 15:32:45 DEBUG[9086] channel.c: Scheduling timer at 160 sample intervalsOct 20 15:32:45 VERBOSE[9086] logger.c: -- Playing 'letters/s' (language 'en')Oct 20 15:32:45 DEBUG[5387] chan_iax2.c: Ooh, voice format changed to 4Oct 20 15:32:46 DEBUG[9086] channel.c: Scheduling timer at 36 sample intervalsOct 20 15:32:46 DEBUG[9086] channel.c: Scheduling timer at 0 sample intervalsOct 20 15:32:46 DEBUG[9086] channel.c: Scheduling timer at 0 sample intervalsOct 20 15:32:46 VERBOSE[9086] logger.c: -- Executing Set("IAX2/204.11.194.34:4569-4", "TIMEOUT(digit)=3") in new stackOct 20 15:32:46 VERBOSE[9086] logger.c: -- Digit timeout set to 3Oct 20 15:32:46 VERBOSE[9086] logger.c: -- Executing Set("IAX2/204.11.194.34:4569-4", "TIMEOUT(response)=4") in new stackOct 20 15:32:46 VERBOSE[9086] logger.c: -- Response timeout set to 4Oct 20 15:32:46 VERBOSE[9086] logger.c: -- Executing BackGround("IAX2/204.11.194.34:4569-4", "custom/IsRecordedMsg") in new stackOct 20 15:32:46 DEBUG[9086] channel.c: Scheduling timer at 160 sample intervalsOct 20 15:32:46 VERBOSE[9086] logger.c: -- Playing 'custom/IsRecordedMsg' (language 'en')Oct 20 15:32:49 DEBUG[5387] chan_iax2.c: Immediately destroying 4, having received hangupOct 20 15:32:49 DEBUG[9086] channel.c: Scheduling timer at 0 sample intervalsOct 20 15:32:49 VERBOSE[9086] logger.c: == Spawn extension (ivr-2, s, 8) exited non-zero on 'IAX2/204.11.194.34:4569-4'Oct 20 15:32:49 VERBOSE[9086] logger.c: -- Executing Hangup("IAX2/204.11.194.34:4569-4", "") in new stackOct 20 15:32:49 VERBOSE[9086] logger.c: == Spawn extension (ivr-2, h, 1) exited non-zero on 'IAX2/204.11.194.34:4569-4' -snip-from IAX.conf[didww]username=didwwSecret= (secret changed by todd)Type=userHost=204.11.194.34Insecure=veryContext=ivr-2;Context=from-trunkfrom extensions_additional.conf[ivr-2]include = ivr-2-custominclude = ext-findmefollowinclude = ext-localinclude = app-directoryexten = h,1,Hangupexten = s,1,Noop(${CALLERID})exten = s,n,Noop(${EXTEN})exten = s,n,Set(LOOPCOUNT=0)exten = s,n,Set(__DIR-CONTEXT=default)exten = s,n,Answer; exten = s,n,Set(DID=${EXTEN})exten = s,n,Wait(1)exten = s,n,SayAlpha(${EXTEN})exten = s,n(begin),Set(TIMEOUT(digit)=3)exten = s,n,Set(TIMEOUT(response)=4)exten = s,n,Background(custom/IsRecordedMsg)exten = hang,1,Playback(vm-goodbye)exten = hang,n,Hangupexten = t,1,Goto(ext-queues,200,1)exten = i,1,Playback(invalid)exten = i,n,Goto(loop,1)exten = loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])exten = loop,n,GotoIf($[${LOOPCOUNT} 2]?hang,1)exten = loop,n,Goto(ivr-2,,begin)exten = fax,1,Goto(ext-fax,in_fax,1); end of [ivr-2]On Oct 20, 2006, at 5:48 PM, Eric "ManxPower" Wieling wrote:There is no difference between an extension and a DID as far as Asterisk is concerned. You must have typoed the above example as you do not have an exten = s,1When you do a exten = s,n,SayAlpha(${EXTEN}) the extension IS "s". If it was not "s" then it would never have gotten to that extension.__
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Re: [asterisk-users] getting DID info..
When dialing from internal extension, it gives me the number I dialed
( in my case..). When dialing from AIX or SIP from didww.com
nothing comes through on the SayAlpha thanks for the thought
though... What I want is the number that the user dials to get my
serverI'm also going to look into the ${SIP_HEADER()} variable.
-t-
On Oct 20, 2006, at 7:37 PM, Lacy Moore - Aspendora wrote:
If 's' is the correct extension, as I expect it is, how do I get
the DID number that the call came in on?
What does ${DNID} give you?
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Re: [asterisk-users] Re: 1.4 on mac OSX 10.4.8
I'm a Certified Apple Sys Admin - lots of experience with Macs and Mac servers. However, when setting up an asterisk server, I'm still thinking a Dell box with linux is the best direction - to get the full reliability and full support of this group. Am I mistaken? Or is using a Mac box just as convenient and reliable? Or is traditional linux 'strongly' recommended for asterisk? I'm looking at a solely IP based system - no digium cards thanks Todd ___ --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] DID failover
I'm setting up an asterisk server where an administrator will not always be available in case of problems. While I expect problems to be rare, I need to be prepared. We're thinking of VoIP DID's and SIP phones so it's an all TCP/IP network. We could get a second server to substitute - What is involved in 'transferring' or 're- registering' the DID incoming lines to a second server in case the primary is down? If there a better fall-over method? I'm looking for the easiest way for the un-educated sys-admin-apprentice to handle it. The system doesn't exist yet so any suggestions are appreciated. I recognize I'll need to modify the SIP phones- I'll figure that out later. thanks in advance Todd ___ --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
