Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 09:52 -0400, Dan Journo wrote: Hi Bob, Thanks for that. Is there any way I can make the task run in the background and free up the console? Also so that I can disconnect my ssh session without losing the task. Thanks Dan Matthieu NICAISE mentioned screen which should work. Another way would be to activate the script through cron: 1. create a script that does a few pings and e-mails the results. 2. activate the script with cron as often as needed. Once this is setup, you can quit your ssh access to the remote server. Contact me offlist if you need more information. Best regards, Bob Smither -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: snip How can i log a continuous ping test to a file and include the date and time of each ping? Try this: #!/bin/sh for (( ; ; )) do NOW=$(date +%T %m/%d/%Y) PING=$(ping -qc 1 example.com) echo $NOW: $PING pinger.log done exit 0 You can then monitor the log file using: $ tail -f pinger.log You will need to use ^C to kill the script. Hope this helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
On Fri, 2009-12-18 at 23:56 -0600, Steve Johnson wrote: If you try just this, what does the caller hear? It should be ringing for the first 20 sec, and then maybe the congestion tone afterwards. exten = s,1,Wait(20) exten = s,n,Hangup Dialplan: [cci] exten = s,1,Wait(10) exten = s,n,Hangup() When the number is dialed, here is the CLI output: Connected to Asterisk 1.4.21.1 currently running on k6-2 (pid = 6283) Verbosity was 0 and is now 3 -- Executing [8772709...@inbound:1] Goto(SIP/smither-03390860, cci|s|1) in new stack -- Goto (cci,s,1) -- Executing [...@cci:1] Wait(SIP/smither-03390860, 10) in new stack -- Executing [...@cci:2] Hangup(SIP/smither-03390860, ) in new stack == Spawn extension (cci, s, 2) exited non-zero on 'SIP/smither-03390860' The caller hears silence for 10 seconds. When the Hangup is executed, as reported on the CLI, the caller _then_ hears ringing (!?) which continues until the caller hangs up. Here is the entry in sip.conf (Asterisk registers with the provider): [vitel-inbound-cci] type=friend dtmfmode=auto host=provider host context=inbound username=user name secret=my secret allow=all insecure=very nat=yes Context in extensions.conf: [inbound] exten = 8772709688,1,Goto(cci,s,1) The context [cci] is shown above. I appreciate the help, as I am confused! -- Bob Smither, PhD Circuit Concepts, Inc. = There are only 10 kinds of people in the world --Those who understand binary, and those who don't... = smit...@c-c-i.com http://www.C-C-I.Com 281-331-2744(office) -4616(fax) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
On Sat, 2009-12-19 at 08:26 -0500, cov...@ccs.covici.com wrote:
I have a strange suggestion -- have one extension answer the call and
dial the extension you want -- then it should ring before dialing the
second one.
Actually, that is pretty close to what I do on a *1.6 box and it works.
Here's what I tried on my *1.4 box (in extensions.conf):
[inbound]
exten = 8772709688,1,Dial(Local/s...@cci,15)
exten = 8772709688,n,Hangup()
[cci]
exten = s,1,Set(CallerContext=${CONTEXT}) ; capture context
; document time of call to console
exten = s,n,NoOp(Time is: ${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
; document caller id to console
exten = s,n,NoOp(CallerID is ${CALLERID(all)})
exten = s,n,Set(TIMEOUT(digit)=3) ; Set Digit Timeout
exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout
; create unique call id for this call
exten = s,n,Set(GLOBAL(cid)=${EPOCH})
;
;exten = s,n,Playtones(ring)
exten = s,n,Wait(10)
;exten = s,n,StopPlaytones()
exten = s,n,Answer()
exten = s,n(start),Wait(0.5)
exten = s,n,BackGround(cci/prompt00)
exten = s,n,WaitExten ; Wait for an extension to be dialed.
I tried both with and without the Playtones(ring) / StopPlaytones()
lines.
Here is what I get from the CLI:
Connected to Asterisk 1.4.21.1 currently running on k6-2 (pid = 8998)
Verbosity was 0 and is now 3
-- Executing [8772709...@inbound:1] Dial(SIP/smither-173b4940,
Local/s...@cci|15) in new stack
-- Called s...@cci
-- Executing [...@cci:1] Set(Local/s...@cci-7c61,2,
CallerContext=cci) in new stack
-- Executing [...@cci:2] NoOp(Local/s...@cci-7c61,2, Time is:
2009-12-19 09:43:10) in new stack
-- Executing [...@cci:3] NoOp(Local/s...@cci-7c61,2, CallerID is
* *) in new stack
-- Executing [...@cci:4] Set(Local/s...@cci-7c61,2, TIMEOUT(digit)=3)
in new stack
-- Digit timeout set to 3
-- Executing [...@cci:5] Set(Local/s...@cci-7c61,2,
TIMEOUT(response)=10) in new stack
-- Response timeout set to 10
-- Executing [...@cci:6] Set(Local/s...@cci-7c61,2,
GLOBAL(cid)=1261237390) in new stack
== Setting global variable 'cid' to '1261237390'
-- Executing [...@cci:7] PlayTones(Local/s...@cci-7c61,2, ring) in
new stack
-- Executing [...@cci:8] Wait(Local/s...@cci-7c61,2, 10) in new stack
-- Executing [...@cci:9] StopPlayTones(Local/s...@cci-7c61,2, ) in
new stack
-- Executing [...@cci:10] Answer(Local/s...@cci-7c61,2, ) in new
stack
-- Executing [...@cci:11] Wait(Local/s...@cci-7c61,2, 0.5) in new
stack
-- Local/s...@cci-7c61,1 answered SIP/smither-173b4940
-- Executing [...@cci:12] BackGround(Local/s...@cci-7c61,2,
cci/prompt00) in
new stack
-- Local/s...@cci-7c61,2 Playing 'cci/prompt00' (language 'en')
This all looks as expected to me, but the caller hears nothing until the
BackGround statement is executed. There still is no ringing back to the
caller.
Thanks!
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[asterisk-users] Ringing for incoming call
Dear All, I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by Vitelity. When the number is called it goes to my Asterisk box. The protocol is SIP. This all works just fine if I answer the call and begin a playback. I want to let the number ring for a few seconds before it is answered, and would like the caller to hear it ringing. I have tried: ... exten = s,n,Answer exten = s,n,Playtones(ring) exten = s,n,Wait(10) exten = s,n,StopPlaytones() exten = s,n,BackGround(sound file) ... also ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. In all cases when I call the number I never hear it ringing. After the 10 second delay, the BackGround app does run. Connecting to the CLI does not give me any useful information - for example the Ringing app is shown to run, but the caller does not hear it. Any suggestions? Many thanks! -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote: Try putting the wait before the Answer. ... exten = s,n,Wait(10) exten = s,n,Answer ... Thanks Steve. I tried that: On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither smit...@c-c-i.com wrote: Dear All, snip ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. snip i.e., after the Wait, but still no joy. Anything else I need to look at? Thanks, -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!
On Fri, 2009-12-18 at 19:43 -0800, Ira wrote: At 03:14 PM 12/18/2009, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ And any plan for Skype for Asterisk? Ira http://www.digium.com/en/products/software/skypeforasterisk.php (but it is not free) -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting multiple phones to ring ...
On Mon, 2009-12-14 at 11:49 -0600, Bob Smither wrote: This has to be easy, but I have spent a fair amount of time looking for a solution to no avail. I am trying to get multiple phones to ring when a call comes into an Asterisk box from a particular phone number. What happens is that only one of the phones rings. I have several GrandStream BT200 SIP phones connected via an internal LAN to Asterisk 1.6.0.17 on CentOS. Here is a snippet from sip.conf: [bt112] type=friend host=129.168.0.112 context=bt200 [bt116] type=friend host=192.168.0.116 context=bt200 snip I got this to work by assigning a secret to each phone in sip.conf (and in each phone). Not sure why that worked. Before (with no assigned secret) one phone would ring and the other would not. I could never discern a pattern. Since assigning secrets to each phone, both phones now appear to consistently ring as expected. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting multiple phones to ring ...
This has to be easy, but I have spent a fair amount of time looking for
a solution to no avail. I am trying to get multiple phones to ring when
a call comes into an Asterisk box from a particular phone number. What
happens is that only one of the phones rings.
I have several GrandStream BT200 SIP phones connected via an internal
LAN to Asterisk 1.6.0.17 on CentOS. Here is a snippet from sip.conf:
[bt112]
type=friend
host=129.168.0.112
context=bt200
[bt116]
type=friend
host=192.168.0.116
context=bt200
I am trying to use these phones in extensions.conf (incoming calls are
sent to this context in extensions.conf):
[lrcf]
exten = s,1(start),Set(CallerContext=${CONTEXT}) ; capture context
; document time of call to console
exten = s,n,NoOp(Time is: ${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
; document caller id to console
exten = s,n,NoOp(CallerID is ${CALLERID(all)})
exten = s,n,Set(TIMEOUT(digit)=3) ; Set Digit Timeout to 3 seconds
exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout
; create unique call id for this call
exten = s,n,Set(GLOBAL(cid)=${EPOCH})
exten = s,n,Wait(0.5)
exten = s,n,Playback(lrcf/connecting-your-call)
; try IP phones for 10 seconds
exten = s,n,Dial(SIP/bt112SIP/bt116,10,d)
Here is the console output when a call comes in:
-- SIP/-inbound-lrcf- Playing
'lrcf/connecting-your-call.slin' (language 'en')
-- Executing [...@lrcf:9] Dial(SIP/-inbound-lrcf-,
SIP/bt112SIP/bt116,10,d) in new stack
== Using SIP RTP CoS mark 5
-- Called bt112
== Using SIP RTP CoS mark 5
-- Called bt116
-- SIP/bt116-087d1308 is ringing
As indicated, the 'connecting-your-call' message is played and can be
heard by the incoming caller. Although phone bt112 is called, it does
not ring. The phone bt116 is called and it does ring.
I have set the call-limit parameter to 10 within the [general] context
in sip.conf.
What am I missing?
Many thanks!
--
Bob Smither, PhD Circuit Concepts, Inc.
=
recursive, adj. - See 'recursive'
=
smit...@c-c-i.com http://www.C-C-I.Com 281-331-2744(office) -4616(fax)
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Re: [asterisk-users] Problem compiling Zaptel
On Sun, 2008-07-13 at 22:12 -0400, Noah Miller wrote: Hi Bob - snip Are you using any Xorcom hardware? If not, you can avoid this issue by disabling the appropriate items when you run make menuselect before compiling Zaptel. Thanks! That allowed me to compile. Regards, -- Bob Smither [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem compiling Zaptel
Dear All, I have a problem compiling Zaptel on an up to date CentOS 5.2 box. Zaptel 1.4.11, CentOS running on AMD dual core X64. The configuration step finishes, but during the 'make' step it stops here: ... CC [M] /projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /projects/asterisk/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /projects/asterisk/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here make[4]: *** [/projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 ... I am not sure how to correct this. Any suggestions? Thanks! -- Bob Smither [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Woes
On Wed, 2007-12-26 at 17:47 +0200, Tzafrir Cohen wrote: On Wed, Dec 26, 2007 at 08:38:31AM -0600, Bob Smither wrote: On Wed, 2007-12-26 at 07:27 +0200, Tzafrir Cohen wrote: Just to stress the point: you mention that you don't see the card on lspci. If this is so: the problem is not with Zaptel - it is below. Zaptel doesn't even get a chance. Thanks Tzafrir. The last thing I tried was to downgrade Zaptel and during the boot, Zaptel reported success(*). What version of Zaptel reported success? Which version has failed? After some frustrating times, I began to suspect the cards. Doing more careful testing, keeping track of the cards, confirmed that two of four cards I was trying had problems. I am able to get the two working cards to be properly configured and they seem to work well. Thanks for the trouble shooting suggestions from this list! One last question if someone has an answer - Using Kwelstart signaling * can detect incoming calls and can detect hang ups, but doesn't seem to be able to detect when an outgoing (over a Zap channel) call is answered. Is this correct, or have I not configured something? I am comparing this to what I have seen with IAX channels. When using call files, when the IAX channel answers that event in noted on the CLI. I don't see this with the Zap channel using call files. Thanks! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Woes
On Wed, 2007-12-26 at 07:27 +0200, Tzafrir Cohen wrote: Just to stress the point: you mention that you don't see the card on lspci. If this is so: the problem is not with Zaptel - it is below. Zaptel doesn't even get a chance. Thanks Tzafrir. The last thing I tried was to downgrade Zaptel and during the boot, Zaptel reported success(*). Using lspci -v does show the card - however, a cat /proc/interrupts does not show any interrupt assigned to it and if I run ztcfg the machine hangs. * - Another difference may be that I powered the machine off before booting rather than simply restarting (which I had done before). I suspect that was actually the significant difference between attempts. It was late so I stopped - will further investigate this today. Thanks for all the help from this list. -- Bob Smither, PhD Circuit Concepts, Inc. = Grief is not a sign of weakness, nor lack of faith. It is the price we pay for Love --Darcie Sims = [EMAIL PROTECTED] http://www.C-C-I.Com 281-331-2744(office) -4616(fax) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Woes
On Wed, 2007-12-26 at 15:46 +1000, Mattt wrote: Sounds like a PCI bus version issue ;-) Thanks Matt, but could you elaborate? The card supposedly supports both 5 and 3.3 volt PCI slots, and the slots on the older system that works with the card appear the same as on the board giving me grief. The connectors in both systems have a single key while the card has two keys. An further explanation of your comment would be appreciated! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Woes
On Wed, 2007-12-26 at 10:22 -0500, dave cantera wrote: bob, look on p20 of 'the book' edition 2, or p16 edition 1 this explains the 3.3v vs 5.0v issue with motherboard slots http://www.oreilly.com/catalog/9780596510480/ daveC The slots are 5V PCI on both motherboards. The X100P cards (ordered from X100P.com) have two keys and are advertised to work in both 5 and 3.3 V slots. Thanks, -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Woes
On Tue, 2007-12-25 at 09:40 +0200, Tzafrir Cohen wrote: On Tue, Dec 25, 2007 at 01:02:25AM -0600, Bob Smither wrote: On Mon, 2007-12-24 at 22:19 -0500, Alex Balashov wrote: Does it show up in 'lspci'? No - With PnPOS = yes, card in first PCI slot - lspci -v does not show the card. There are several unused interrupts. So until you see it there, no point in wasting your time on drivers. Can you boot from a different kernel and check if you see the card? Try various kernel / BIOS options? I think I have tried all the BIOS options. I believe I will try an older Zaptel version and if that doesn't work simply start from scratch and use the CentOS 4.4 / older Zaptel that does work on another (AMD) box. Many thanks for the feedback. Another list member posted privately that he had similar problems, but did find an older Zaptel that works on CentOS 5.1 I will let the list know if I find a solution. -- Bob Smither, PhD Circuit Concepts, Inc. = Whenever a man has cast a longing eye on them (public offices), a rottenness begins in his conduct - Thomas Jefferson = [EMAIL PROTECTED] http://www.C-C-I.Com 281-331-2744(office) -4616(fax) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100P Woes
I am out of things to try on getting an X100P card to work. I have a P4 system with CentOS 5.1 installed (network install). After adding required packages I compiled (apparently successful) and installed Zaptel in anticipation of installing Asterisk. I simply cannot get the X100P card to be recognized. I have tried bios setting of PnPOS no and PnPOS yes, tried the card in different slots, disabled all the unused motherboard devices to free up interrupts, etc. The card works flawlessly in a CentOS 4.4 install on an old K6-2 motherboard. If I put an ethernet card in a PCI slot (in addition to the one on the motherboard), it is recognized and configured during boot so I assume the interrupt hardware is working on the mother board. I would really appreciate any suggestions of what I could try here. Thanks! -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Woes
On Mon, 2007-12-24 at 22:19 -0500, Alex Balashov wrote: Does it show up in 'lspci'? No - With PnPOS = yes, card in first PCI slot - lspci -v does not show the card. There are several unused interrupts. I should have included earlier: Zaptel 1.4.7.1 Thanks. -- Bob Smither, PhD Circuit Concepts, Inc. = Foreign aid might be defined as a transfer from poor people in rich countries to rich people in poor countries. --Douglas Casey (1992) = [EMAIL PROTECTED] http://www.C-C-I.Com 281-331-2744(office) -4616(fax) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback file and detect a key press
On Sat, 2007-12-08 at 13:40 -0600, Moises Silva wrote: Bob, GET DATA should do something like that. But, to do exactly that, you could try a patch I did to call AGI(agi:async), this is a special way of AGI. As you know, you can already call AGI(name-of-script.php), or AGI(agi://ipaddress) for the so named FastAGI. I created another variant AGI(agi:async) that will allow you to run AGI thru the manager interface. That way you could easily do this in a language like php. Thanks Moises - I will take a look at your patch, it sounds interesting. I found a developed a workaround using a standard AGI script. The message is played back using Stream File with 0 offset and the allowed escape digits. When an allowed digit is pressed, the digit and the file offset is returned by Stream File. Using the pressed digit you can go to an appropriate routine to play back an acknowledgment Thank you for pressing ... and then start the message playback just before the time that the key was pressed. The user hears the message, then has feedback that the key was detected, and then hears the rest of the message. This actually works pretty well. Thanks! -- Bob Smither, PhD Circuit Concepts, Inc. = You have the right to YOUR life, liberty and property, but not MINE = [EMAIL PROTECTED] http://www.C-C-I.Com 281-331-2744(office) -4616(fax) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playback file and detect a key press
I would like to do the following: Play back a file, and during the playback be able to detect a DTMF tone that may be pressed. I do not want to interrupt the playing of the file, but when the file finishes I would like to be able to tell if a key was pressed and which key it was. Anyway to do this? In AGI: o Wait for Digit waits for a digit to be pressed, and I don't see how to play a file at the same time. o Stream File can detect a digit, but then the file playback is interrupted. In a call plan: o Playback plays a file but does not detect pressed digits. o Background plays a file, but stops the playback when a key is pressed. Is there anyway to do what I want to do? Thanks! -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what version is running?
On Sat, 2007-04-14 at 06:23 -0400, Joe Acquisto wrote: A very simple question - what version is running? A CLI - show version does not tell me, only shows info about who compiled, and when. A google and other nefarious devices have not yielded the secret. Here is what worked for me: ast2*CLI core show version Asterisk 1.4.2 built by root @ ast2.c-c-i.com on a x86_64 running Linux on 2007-04-08 01:31:11 UTC ast2*CLI HTH -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring audio file legth
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote: Hi, I have to set call length to 3min, but before hangup have to warn caller. There are many IVRmenu and submenu options with different warning audio. I have to measure somehow the audio file length and subtract it from 3 minutes. I have not tried this, so I may be off - but do you really have to do this? The documentation I have indicates that if there is an extension T in the context, that extension is used at the absolute timeout. So, would: exten = T,1,play your warning message exten = T,n,Hangup work? HTH, -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Gain Settings
On Sat, 2007-04-07 at 23:52 -0500, Eric ManxPower Wieling wrote: snip The device doing the IP/TDM conversion should be the device that sets the gains correctly. The same applies to echo canceling. As I stated, this started with the warning of Novice Question :-). Eric, can you elaborate on the above? Is the device you are referring to within Asterisk or somewhere else in VOIP land? I am not sure what to do with this information. If it matters - the clipping behavior I see is in voices recorded on Asterisk 1.4.2 from a call placed over Packet8 and routed back to my Asterisk box through NuFone.net. Same happens from a POTS call routed back to my Asterisk box through NuFone.net. Thanks, -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Audio Gain Settings
On Fri, 2007-04-06 at 21:42 -0700, Yuan LIU wrote: snip You are right. zapata.conf is not used in IAX connections. My reading has led me to believe that manipulating gain on an IP PBX is neither necessary nor practical in VoIP channels, so Asterisk does not devise such settings. Thanks Yuan. I beg to differ with the developers if there really is no amplitude control on IP channels. I have an application where I am studying the spectrum of recorded voice. When I call into my Asterisk box I have to hold the phone away and speak softly to avoid clipping the recorded waveform - clipped waveforms play havoc with the spectrum. I guess it is time to study the source code (ugh!). Best regards, -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio Gain Settings
Warning - novice question ahead! Dear List, I have installed Asterisk 1.4.2 on an AMD dual core x86-64 box running CentOS 4.4. Compilation and installation were straightforward. The box only supports IAX connections so I have no zap hardware. My question is this - where do I set the txgain and rxgain parameters for the IAX channels? With a previous setup I used settings in zapata.conf, but I believe these are not used with the IAX connections (?). Thanks for any insight. -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
