Re: [asterisk-users] Vicibox vs VicidialNow
ViciBox actually gives you the option of using the 2.2.1 release or SVN/trunk versions ViciDial Also, ViciBox is the officially supported ISO installer of the ViciDial project. But, both ViciBox and ViciDialNow are Linux ISO installers that will give you a functional ViciDial system. Thanks, MATT--- On Sun, Jul 25, 2010 at 8:29 PM, Juan David Diaz juanch...@gmail.comwrote: The only big difference I know, is: VicidialNow - *based on CentOS* - Vicidial 2.0.5.1rc1 ViciBox - *Based on OpenSuse* - Vicidial 2.0.5 The core of the call center for both of them is Vicidial. Regards. 2010/7/25 Alejandro Cabrera Obed aco1...@gmail.com Dear all, I need a call center asterisk's based solution and I see there are two important solution for 120+ agents: VicidialNow and ViciBox Can you tell me the difference between these open source call center solution please ??? Special thanks Alejandro -- _ -- 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 -- Juan. Linux User #441131 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio using xlite
On Sun, Jul 25, 2010 at 10:20 PM, Janu Mukherjee janu.mu...@gmail.com wrote: I installed asterisk server in my linux box. I configured a user 1000 using xlite and registered with asterisk server in the same linux box. I Where on the network is this box? configured one more user 1001 in other box and this user also got registered with asterisk. But i am facing two issues here. Where on the network is this other box? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adit 600 over MGCP.
Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the slowsequence = yes line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found online.) No improvement. A working configuration or any other tips and ideas regarding this would be most welcome. Thank you in advance. Magnus Persson Log output: Jul 21 13:59:18 crabbofix asterisk[2689]: NOTICE[6220]: chan_mgcp.c:3564 in mgcp_request: Asked to get a channel of unsupported format '0' Jul 21 13:59:18 crabbofix asterisk[2689]: NOTICE[6220]: channel.c:2901 in __ast_request_and_dial_uniqueid: Unable to request channel MGCP/aal n/3...@adit.westel.nt and Jul 21 16:11:01 crabbofix asterisk[2689]: NOTICE[6263]: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.30.10 mgcp.conf (relevant parts) [general] port = 2427 ;port = 2727 bindaddr = 192.168.30.1 disallow = all allow = alaw [adit.westel.nt] host = 192.168.30.10 context = westel callgroup = 0 pickupgroup = 0 transfer = no cancallforward = no canreinvite = no dtmfmode = inband ;dtmfmode = rfc2833 wcardep=* ; define the internal lines callerid = Westel 069022130 line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 ; remove all features transfer = no cancallforward = no canreinvite = no context = newmrg ;dtmfmode = rfc2833 dtmfmode = inband ; define old TeleVaxel callerid = MRG 2150 line = aaln/5 line = aaln/6 line = aaln/7 (snip)... Adit configuration (relevant parts): adit print config - -Adit 600 configuration file -Created on 01/24/2002 at 02:32:23 -This file is valid for the following configuration only: - -CardType - -SLOT A ITE SW Version: 9.0.0 -SLOT 1 FXS8Ax8 -SLOT 2 FXS8Ax8 -SLOT 3 FXS8Ax8 -SLOT 4 FXS8Ax8 -SLOT 5 FXS8Ax8 -SLOT 6 CMGx1 (snip)... -Setting slot 6 CMG. (snip)... set 6:1 up set 6 snmp name unknown set 6 snmp contact unknown set 6 snmp location unknown set 6 ntp server 192.168.30.1 set 6 ntp timezone 1 set 6 ntp enable set 6 cdr enable set 6 hookflash 0 set 6 mgcp addressformat nobrackets set 6 mgcp callagent address 192.168.30.1 set 6 mgcp callagent port 2427 set 6 mgcp up set 6 mgcp rsipwildcard enable set 6 voip ptime g711mu 10 set 6 voip ptime g711a 10 set 6 voip rtcp cname adit set 6 compander alaw set 6 voip sdpaddress gatewayid set 6:1:1:1 log start mgcp set 6:1:1:1-48 algorithm preference g711mu g711a g726_16 g726_24 g726_32 \ g726_40 set 6:1:1:4 echo cancellation disable set 6:1:1:4 fax bypass set 6:1:1:4 modem bypass reset 6 -Turning verification on. set verification on adit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Vertical IP2007 phones with Asterisk?
From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Sunday, July 25, 2010 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Kevin Keane Subject: Re: [asterisk-users] Using Vertical IP2007 phones with Asterisk? Kevin Keane wrote: I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I would like to use the phones with an Asterisk system instead, but there doesn't seem to be much information on it on Google. Is it even possible? These phones claim that they are SIP phones. Thanks! Kevin Good luck getting much help from Vertical. For those who don't know, Vertical seems to have bought all the semi-crappy business phone system companies in recent years, Comdial, Vodavi (who was previously merged with Isoetec and Executone ) and perhaps others. The only decent product in the mix was the Keyvoice voice mail line. Previous dealers seem to have been shut out of support, and there is little available without being a Vertical dealer. Seems they may be using the Cisco model. Either try and sign up with Vertical, or put the system on eBay and cut your losses John Novack I have done some more research and found that it is indeed possible to use these phones with Asterisk - at least, they register (I haven't fully tested everything yet). I have tried it with Firmware version 3.2.32 - G729. Firmware updates unfortunately are only available through Vertical dealers. In case somebody else uses Google to find it, here is what I found out. The phones, as far as I can tell, do not support DHCP option 66. It does support a configuration file on a TFTP server, but it will only load the configuration file when you tell it to do so in the Web interface. It might be possible to automate that with a wget script. To create the configuration file, it is best to start by manually configuring one phone with a Web browser. After a factory reset, the user name/password are admin/1234. The connection to Asterisk is configured in the SIP tab. The Asterisk server goes into the Registrar Server and Registrar Outbound Server fields. It MUST be an IP address; the phone does not accept a DNS name here. Phone Number and Authorized ID should be the extension. The Phone number will be displayed on the phone's screen. The User name will also be displayed; you can type what you want here, but be aware that you cannot use a space! The secret (Authorized Password) also accepts only certain characters; it is best to stick to just alphanumeric and avoid punctuation altogether. Once you are done, you can save the configuration to a file. In the SW Upgrade tab, click on Download Settings. Move the downloaded file to your TFTP server and rename it IP2007.cfg . Edit as appropriate for the next phone. It is a text file. Be sure to adhere to the same restrictions as the Web interface. Otherwise, the file will not load, and there is no indication what is wrong. Then log on to the next phone's Web interface. Go to the SW Upgrade page. Change the IP address of the TFTP server (no DNS names accepted). Click on Save (otherwise, the phone will use the previous setting for the TFTP server!). Make sure the file name under Profile is correct, and click on the Update button next to it. I hope this helps the next person trying these phones! -- _ -- 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] 1.6.2.10 sounds Makefile error?
On Sun, Jul 25, 2010 at 08:06:55PM +0100, Faris Raouf wrote: I'm having some mysterious problems installing 1.6.2.10 on Centos 4.8 (totally up to date). I can't see anything on Google or the list regarding this issue, which I find a bit odd considering 1.6.2.10 was released a few days ago. I'm therefore assuming there's something weird about my setup, even though there shouldn't be! I had no problems with 1.6.2.7 or any other release. It is just 1.6.2.10 that's causing the problem. I've tried using an svn checkout and downloading asterisk-1.6.2.10.tar.gz and asterisk-1.6.2-current.tar.gz but the same thing happens. Basically, after the usual ./configure and make, when I make install I get the following (this is from the SVN attempt but other than the paths all is the same): #make install CFLAGS= -I/usr/include/libxml2 -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -march=i686 build_tools/mkpkgconfig /usr/lib/pkgconfig; mkdir -p /var/lib/asterisk/static-http for x in static-http/*; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/static-http ; \ done if [ -d doc/tex/asterisk ] ; then \ mkdir -p /var/lib/asterisk/static-http/docs ; \ for n in doc/tex/asterisk/* ; do \ /usr/bin/install -c -m 644 $n /var/lib/asterisk/static-http/docs ; \ done \ fi mkdir -p /var/lib/asterisk/images for x in images/*.jpg; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/images ; \ done mkdir -p /var/lib/asterisk/agi-bin make -C sounds install make[1]: Entering directory `/root/asterisk-svn/asterisk-1.6.2/sounds' Makefile:144: *** missing separator. Stop. make[1]: Leaving directory `/root/asterisk-svn/asterisk-1.6.2/sounds' make: *** [datafiles] Error 2 I don't have such a centos 4.8 system handy to test with. What version of 'make' do you have? make --version rpm -q make In any case, please submit a report to http://issues.asterisk.org/ Looking at the sounds directory, I have Makefile and sounds.xml Running make install in that directory gives me the same Makefile:144 *** missing separator. Stop When I copy across the sounds/Makefile from my 1.6.2.7 source directory to the 1.6.2.10 source directory, all is well again and I can make install with no errors. I did a diff on the two Makefiles and there are what appear to be several differences, but I can't put my finger on any obvious errors. My initial suspect is http://svnview.digium.com/svn/asterisk?view=revisionrevision=267820 as it uses some not-completely-standard Makefile directives. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreeTDS (Microsoft MsSQL 2008) and CDR
Hi, I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, freetds-bin, but, when I run configure and then make menuconfig in section Call Detail Recording - cdr_tds it's disabled. It only writes that Depends on: freetds(E). On another server (same configuration) I installed the same packages, and it's working fine. Any suggestions, what I did wrong? Regards Andraž attachment: cdr.PNG-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_skinny still maintained?
Hi, I've managed to acquire a few Cisco handsets (7905, 7920) and would like to use them with Asterisk. Rather than simply switching to the SIP firmware I thought I'd use these with chan_skinny - partly because this is Cisco's primary firmware and therefore the phones might be more stable, and partly to help test chan_skinny as it seems to be generally underused. (Is functionality identical across both firmwares?) However, I've come across a couple of showstoppers and am not really sure where to go from here. I've raised bugs for both of them (#17680, #17692) and had no response so far - have I perhaps overestimated how much chan_skinny is in use these days, or do I need to follow another route? I'm not an Asterisk developer but am happy to spend some time this week resolving the problems. Unfortunately I need the phones next week, so may have to end up taking the defeatist approach of switching to the SIP firmware :( Cheers! Jonathan -- If we knew what it was we were doing, it would not be called research, would it? - Albert Einstein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No audio using xlite
Hi, I installed asterisk server in my system running linux. I configured a user 1000 using xlite and registered with asterisk server in the same linux system. I configured one more user 1001 in another linux machine and this user also got registered with asterisk. But i am facing two issues here. 1. When a call is made from 1001 to 1000 i could see an incoming call blinking but no audio flow is observed. 2. When i made a call from 1000 to 1001 it is showing incoming on line 3 of 1000. What could be the problem. I wrote the dial plan as follows. [default] exten=1000,1,Dial(SIP/1000) exten=1001,1,Dial(SIP/1001) Can anyone please help me to solve this. Thanks in Advance, Saritha. -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
On Monday 26 Jul 2010, Andraž wrote: Hi, I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, freetds-bin, but, when I run configure and then make menuconfig in section Call Detail Recording - cdr_tds it's disabled. It only writes that Depends on: freetds(E). On another server (same configuration) I installed the same packages, and it's working fine. Any suggestions, what I did wrong? Run $ dpkg -l on both boxes, and compare the output. (Don't forget, you can always do something like $ ssh 10.11.12.13 'dpkg -l' other_box_packages to run a command on another machine and trap its output in a file on yours. Just substitute the appropriate IP address or hostname.) Chances are, there's a -dev package you've missed out. (Why distributions still persist in separating out -dev packages in these days of fast CPUs, broadband internet connections and terabyte hard disks is beyond me, but that's for another day .) -- AJS -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
On Mon, Jul 26, 2010 at 10:05:27AM +0200, Andraž wrote: Hi, I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, freetds-bin, but, when I run configure and then make menuconfig in section Call Detail Recording - cdr_tds it's disabled. It only writes that Depends on: freetds(E). On another server (same configuration) I installed the same packages, and it's working fine. Any suggestions, what I did wrong? Have you re-ron ./configure #? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] URGENT - who picked up the call??
Hello,
I've been looking for this on voip-info and this list threads, and I am
guessing I am not looking right.
What I need is the way to capture (and write to DB) the information on who
'picked' or 'received' the incoming call.
Here is the example of .rb file that is called from extensions.conf:
private
def lokal
call_log_id = nil
begin
call_log_id = call_log()
$agi.answer
$agi.exec('WAIT', '2')
local_channels =
get_locals()
dial_params =
local_channels.join('')
dial_params
||m(moh-0900...@moh_id}) if moh_available?()
1.times do
r =
$agi.exec('DIAL', dial_params)
r =
$agi.get_variable('DIALSTATUS')
retry if
r.message.include?('BUSY')
end
ensure
call_log(call_log_id) unless
call_log_id.nil?
end
end
private
def get_locals
local_channels = @locals.map { |x|
'Sip/operator1Zap/' + x.strip }
# FIX - ovaj raise treba da prijavi
nedefinisane lokale za servis a ne za telefon
raise Nisu definisani lokalni kanali u
settings za telephone_id = #...@settings_row['telephone_id']} if
local_channels.empty?
local_channels
end
As you see the call can be picked either by the Zap channels in locals of
SIP/operator user. Now i Need to know here:
$my.select_db('tvr2')
$my.query(UPDATE call_log
SET endtime = NOW(), local=#SOMETHINGHERE# WHERE id = #{call_log_id})
raise 'Cant write from log:
call_log_id = #{call_log_id}' if $my.affected_rows() != 1
if place of #SOMETHINGHERE# - where the call was transferred (from the part
above).
Anyone?
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[asterisk-users] Optimize peers registration under jitter/delay.
Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using operator Internet with delay/jitter conditions? I chooses values above after many tests but still have some problems: - from time to time peers have lagged connections... maximum time to re register is 10 seconds... can i minimize that or refresh without unregister and re register? i want that users to be as much as he can online even in delay/jitter conditions. Of course if is response time too much like over 1000, 2000 ms i prefer to re register if it can. - is any connection between these timers for keepalive connections , re register etc... and choppy sounds/sometimes interrupted/nosy, in an active call? If yes how can i optimize both things: to hav' a good sound and to keepalive connections for peers. Thank you very much for support... please feel free to ask me any question or misunderstanding of this mail, and I'll email you with more detail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] URgent - capturing 'answered'
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
I already have this:
$my.query(UPDATE call_log
SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE call_log
SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id})
And in above example it would write SIP/operator1-e77f into answeredby.
Any help is greatly appreciated!
--
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Re: [asterisk-users] Optimize peers registration under jitter/delay.
We are having good results with maxexp 120 minexp 90 defexp 100 qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using operator Internet with delay/jitter conditions? I chooses values above after many tests but still have some problems: - from time to time peers have lagged connections... maximum time to re register is 10 seconds... can i minimize that or refresh without unregister and re register? i want that users to be as much as he can online even in delay/jitter conditions. Of course if is response time too much like over 1000, 2000 ms i prefer to re register if it can. - is any connection between these timers for keepalive connections , re register etc... and choppy sounds/sometimes interrupted/nosy, in an active call? If yes how can i optimize both things: to hav' a good sound and to keepalive connections for peers. Thank you very much for support... please feel free to ask me any question or misunderstanding of this mail, and I'll email you with more detail. -- _ -- 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] Proprietary add-ons for Asterisk 1.8
On 07/25/2010 02:47 PM, Richard Kenner wrote: At what stage will there be versions of the G.729 codec, res_cepstal, skypeforasteric, Vestec, etc that'll work with 1.8? It would be good if people using that software could participate in the Beta. We don't normally produce versions of our binary addons for a new release until it reaches the 'release candidate' stage, since up until that time there is the potential for APIs to change that would necessitate rework and rebuilding of the modules. This does mean that some users can't help with beta testing, which is unfortunate, but it's also a way to reduce the burden on our development team during the beta testing period. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URgent - capturing 'answered'
You need to do it by manager interface
Regards,
Faisal Hanif
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
I already have this:
$my.query(UPDATE
call_log SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE
call_log SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE}
WHERE id = #{call_log_id})
And in above example it would write SIP/operator1-e77f into answeredby.
Any help is greatly appreciated!
__ Information from ESET NOD32 Antivirus, version of virus
signature database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
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Re: [asterisk-users] URgent - capturing 'answered'
Hi Faisal,
Isn't it possible to be done via asterisk variables?
I would need it to be done that way - i am sure there's a variable capturing
who answered.
Zarko
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif
Sent: Monday, July 26, 2010 12:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
You need to do it by manager interface
Regards,
Faisal Hanif
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
I already have this:
$my.query(UPDATE call_log
SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE call_log
SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id})
And in above example it would write SIP/operator1-e77f into answeredby.
Any help is greatly appreciated!
__ Information from ESET NOD32 Antivirus, version of virus signature
database 5313 (20100726) __
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Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
If you use cdr_mysql, then this data should already be written to the
dstchannel column in the cdr table.
I already have this:
$my.query(UPDATE call_log
SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE call_log
SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id})
Alternatively you may be able to access ${CDR(dstchannel)}.
I've not checked any of the above, but I believe it is right.
Regards,
Steve
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Re: [asterisk-users] Optimize peers registration under jitter/delay.
did you also hav qualify and qualifyfreq? Thank you for reply, On Mon, Jul 26, 2010 at 1:55 PM, Faisal Hanif fai...@vopium.com wrote: We are having good results with maxexp 120 minexp 90 defexp 100 qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using operator Internet with delay/jitter conditions? I chooses values above after many tests but still have some problems: - from time to time peers have lagged connections... maximum time to re register is 10 seconds... can i minimize that or refresh without unregister and re register? i want that users to be as much as he can online even in delay/jitter conditions. Of course if is response time too much like over 1000, 2000 ms i prefer to re register if it can. - is any connection between these timers for keepalive connections , re register etc... and choppy sounds/sometimes interrupted/nosy, in an active call? If yes how can i optimize both things: to hav' a good sound and to keepalive connections for peers. Thank you very much for support... please feel free to ask me any question or misunderstanding of this mail, and I'll email you with more detail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'dirty' upgrade of 1.4
Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Obviously, I will need to keep my config files (and sound files etc) - so I'll back them up first. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Management interface
I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 Tony -- _ -- 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] URgent - capturing 'answered'
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what is
wrong - if you have any idea please help.
Zarko
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: Monday, July 26, 2010 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
If you use cdr_mysql, then this data should already be written to the
dstchannel column in the cdr table.
I already have this:
$my.query(UPDATE call_log
SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE call_log
SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id})
Alternatively you may be able to access ${CDR(dstchannel)}.
I've not checked any of the above, but I believe it is right.
Regards,
Steve
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[asterisk-users] PBX Lua with Asterisk ODBC
Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce -- _ -- 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] Management interface
On Mon, Jul 26, 2010 at 8:15 AM, Tony LaMear tlam...@indyzoo.com wrote: I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 http://oss.oetiker.ch/mrtg/ -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'dirty' upgrade of 1.4
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- 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] URgent - capturing 'answered'
It gives me headaches trying to use databases with Asterisk. That being said, IMO the best answer to your query is to use the FORKCDR command so that the call will be split into legs. When the operator answers the call, that will be leg 1. When the call is transferred to the desired party, that will be leg 2. Each leg will have it's own CDR entry. -- _ -- 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] URgent - capturing 'answered'
Depending on the version of Asterisk you are running you can call a macro or an
agi as option to dial. These will be called when the line is answered and you
can find the channel name of who answered.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 5:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what is
wrong - if you have any idea please help.
Zarko
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: Monday, July 26, 2010 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
If you use cdr_mysql, then this data should already be written to the
dstchannel column in the cdr table.
I already have this:
$my.query(UPDATE call_log
SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE call_log
SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id})
Alternatively you may be able to access ${CDR(dstchannel)}.
I've not checked any of the above, but I believe it is right.
Regards,
Steve
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
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database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
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Re: [asterisk-users] Optimize peers registration under jitter/delay.
We are not using qualify for the peers which are not on static IP and registering to server. Regards, Faisal Hanif // On 7/26/2010 5:06 PM, Catalin S. wrote: did you also hav qualify and qualifyfreq? Thank you for reply, On Mon, Jul 26, 2010 at 1:55 PM, Faisal Haniffai...@vopium.com wrote: We are having good results with maxexp 120 minexp 90 defexp 100 qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using operator Internet with delay/jitter conditions? I chooses values above after many tests but still have some problems: - from time to time peers have lagged connections... maximum time to re register is 10 seconds... can i minimize that or refresh without unregister and re register? i want that users to be as much as he can online even in delay/jitter conditions. Of course if is response time too much like over 1000, 2000 ms i prefer to re register if it can. - is any connection between these timers for keepalive connections , re register etc... and choppy sounds/sometimes interrupted/nosy, in an active call? If yes how can i optimize both things: to hav' a good sound and to keepalive connections for peers. Thank you very much for support... please feel free to ask me any question or misunderstanding of this mail, and I'll email you with more detail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'dirty' upgrade of 1.4
You should be able to compile the new version, stop asterisk then make install. If you do not do make samples then your conf files will be left alone. Once you have done make install you can the start asterisk again. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 26, 2010, at 5:11 AM, Andrew Thomas wrote: Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Obviously, I will need to keep my config files (and sound files etc) - so I'll back them up first. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] URgent - capturing 'answered'
Hi,
If you need to get it in variable you can read it from dialstatus after
dial and channel-state while ringing.
Regards,
On 7/26/2010 4:14 PM, Zarko Zivanovic wrote:
Hi Faisal,
Isn't it possible to be done via asterisk variables?
I would need it to be done that way -- i am sure there's a variable
capturing who answered.
Zarko
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Faisal
Hanif
*Sent:* Monday, July 26, 2010 12:57 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] URgent - capturing 'answered'
You need to do it by manager interface
Regards,
Faisal Hanif
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
I already have this:
$my.query(UPDATE
call_log SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE
call_log SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE}
WHERE id = #{call_log_id})
And in above example it would write SIP/operator1-e77f into answeredby.
Any help is greatly appreciated!
__ Information from ESET NOD32 Antivirus, version of virus
signature database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
__ Information from ESET NOD32 Antivirus, version of virus
signature database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
__ Information from ESET NOD32 Antivirus, version of virus
signature database 5313 (20100726) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
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Re: [asterisk-users] Management interface
use cacti Regards, Faisal Hanif /Think about the environment before printing this mail /P/ Tænk på miljøet før du printer denne mail/ On 7/26/2010 5:15 PM, Tony LaMear wrote: I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 *Tony * -- _ -- 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] PBX Lua with Asterisk ODBC
you can use all asterisk dial-plan functions and application in lua plus additional complete lua features. so answer is yes. Regards, Faisal Hanif On 7/26/2010 5:34 PM, Bruce McAlister wrote: Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce -- _ -- 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] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a $ as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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Re: [asterisk-users] 'dirty' upgrade of 1.4
Hi Danny, I understand (and welcome) the separate src directories. This would allow me to 'revert' should I feel the need (assuming I can just re-compile over each one). I just need to know if I can re-compile over the existing first. Thanks for your reply. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 26 July 2010 14:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes
On 07/26/2010 11:57 AM, Alexander Aksarin wrote: On 20:59 Fri 23 Jul , Steve Underwood wrote: That's just how your images look for me, so I guess your problem is described here http://www.soft-switch.org/spandsp_faq/ar01s09.html Steve Big thanks for your help, Steve. I tried feh, gqview, gimp and pages look an odd shape. Can you say what image viewer you use for tiff? I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. Steve -- _ -- 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] 1.6.2.10 sounds Makefile error?
I don't have such a centos 4.8 system handy to test with. What version of 'make' do you have? make --version rpm -q make In any case, please submit a report to http://issues.asterisk.org/ Thanks Tzafrir. GNU Make 3.80 Make-3.80-7.EL4 I'll submit a bug report. I just can't figure out why it would only be me seeing this. I'm mystified. Faris. -- _ -- 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] PBX Lua with Asterisk ODBC
Thanks for the quick response, however, how would I access an odbc dsn from the pbx_lua dialplan that has been defined in res_odbc.conf or related odbc structures? I've not come accross any documentation on that feature yet. Any tips/info/links would be appreciated. On 26/07/10 14:33, Faisal Hanif wrote: you can use all asterisk dial-plan functions and application in lua plus additional complete lua features. so answer is yes. Regards, Faisal Hanif On 7/26/2010 5:34 PM, Bruce McAlister wrote: Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce -- _ -- 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] Management interface
On 10-07-26 08:15 AM, Tony LaMear wrote: I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 I've been looking at the OpenNMS project recently. http://www.opennms.org Leif Madsen. -- _ -- 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] PBX Lua with Asterisk ODBC
You need to create a function is res_odbc for each of required query and then u can use that function as normal asterisk dialplan function. Regards, Faisal Hanif On 7/26/2010 7:02 PM, Bruce McAlister wrote: Thanks for the quick response, however, how would I access an odbc dsn from the pbx_lua dialplan that has been defined in res_odbc.conf or related odbc structures? I've not come accross any documentation on that feature yet. Any tips/info/links would be appreciated. On 26/07/10 14:33, Faisal Hanif wrote: you can use all asterisk dial-plan functions and application in lua plus additional complete lua features. so answer is yes. Regards, Faisal Hanif On 7/26/2010 5:34 PM, Bruce McAlister wrote: Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce -- _ -- 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] URgent - capturing 'answered'
Ok Lief, here's an update:
I changed the ruby script line to:
$my.query(UPDATE call_log SET local='${CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
As you suggested, but now in mysql local field i have exactly this written:
${CDR(dstchannel)}
If wasnt changed to a variable, instead it was directly written to mysql.
Any suggestions?
Zarko
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a $
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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[asterisk-users] asterisk distributed device state = res_jabber Versus res_ais
Hello, as I'm looking for a solution (with asterisk 1.6.2) , my investigations leaded to : - res_ais = libais corosync. (each node need to run corosync / aiexec) - res_jabber = libjabber iksemel. (each node need to be connected on an XMPP server) I've been able to make some successful tests with res_ais on 2 servers but got some CPU issues with corosync after some hours of activity. What's the best solution regarding flexibility and stability and real-time exploitation ? I've got the feeling a good (and old) XMPP server will be more reliable than res_ais which seems to be pretty young. Thank you for your help. Mathieu -- _ -- 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] asterisk app_fax, T.30, weird received faxes
On Mon, Jul 26, 2010 at 09:54:24PM +0800, Steve Underwood wrote: On 07/26/2010 11:57 AM, Alexander Aksarin wrote: On 20:59 Fri 23 Jul , Steve Underwood wrote: That's just how your images look for me, so I guess your problem is described here http://www.soft-switch.org/spandsp_faq/ar01s09.html Steve Big thanks for your help, Steve. I tried feh, gqview, gimp and pages look an odd shape. Can you say what image viewer you use for tiff? I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. convert and the rest of imagemagick should handle multi-page tiff (e.g. convert it to PDF). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] 'dirty' upgrade of 1.4
When you run make, it compiles the binaries in the src directory. Once it is done compiling stop asterisk. Running make install will copy the compiled binaries into their respective folders on your system. Then just start asterisk. If you need to revert, stop asterisk, run make install in the old src directory, then start asterisk. Ryan On Mon, Jul 26, 2010 at 9:45 AM, Andrew Thomas a...@datavox.co.uk wrote: Hi Danny, I understand (and welcome) the separate src directories. This would allow me to 'revert' should I feel the need (assuming I can just re-compile over each one). I just need to know if I can re-compile over the existing first. Thanks for your reply. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 26 July 2010 14:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] asterisk app_fax, T.30, weird received faxes
On 07/26/2010 10:55 PM, Tzafrir Cohen wrote: On Mon, Jul 26, 2010 at 09:54:24PM +0800, Steve Underwood wrote: On 07/26/2010 11:57 AM, Alexander Aksarin wrote: On 20:59 Fri 23 Jul , Steve Underwood wrote: That's just how your images look for me, so I guess your problem is described here http://www.soft-switch.org/spandsp_faq/ar01s09.html Steve Big thanks for your help, Steve. I tried feh, gqview, gimp and pages look an odd shape. Can you say what image viewer you use for tiff? I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. convert and the rest of imagemagick should handle multi-page tiff (e.g. convert it to PDF). The main value in converting FAX TIFFs to PDFs (which basically just encapsulates the TIFF file in a PDF wrapper) is that PDF readers generally get the images right. If the average image viewer was not so broken, converting FAXes to PDFs would be less popular. Steve -- _ -- 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] PBX Lua with Asterisk ODBC
On 10-07-26 10:34 AM, Faisal Hanif wrote:
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
So in the dialplan, after you've modified func_odbc.conf you'd be able to do a
query like:
exten = start,1,Set(MyResult=${ODBC_GET_MY_DATA(banana_lollipop)})
How you structure that in pbx_lua I'm not sure, but you create the functions
with func_odbc.conf, which is probably the piece you're missing.
Leif Madsen.
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Re: [asterisk-users] asterisk distributed device state = res_jabber Versus res_ais
On 10-07-26 10:45 AM, Mathieu wrote: Hello, as I'm looking for a solution (with asterisk 1.6.2) , my investigations leaded to : - res_ais = libais corosync. (each node need to run corosync / aiexec) - res_jabber = libjabber iksemel. (each node need to be connected on an XMPP server) I've been able to make some successful tests with res_ais on 2 servers but got some CPU issues with corosync after some hours of activity. What's the best solution regarding flexibility and stability and real-time exploitation ? I've got the feeling a good (and old) XMPP server will be more reliable than res_ais which seems to be pretty young. On Asterisk 1.6.2, your only option for distributing device state is with res_ais. I've used it in a labbing system and it works well -- the caveat is that your machines need to be on a low latency network (i.e. LAN). With Asterisk 1.8 (currently 1.8.0-beta1) you can use XMPP to distribute your device states over the WAN. I've made it work with the Tigase XMPP server. More information about it can be found in the doc/distributed_devstate-XMPP.txt file. Leif Madsen. -- _ -- 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] chan_skinny still maintained?
On 10-07-26 04:03 AM, Jonathan Hunter wrote: However, I've come across a couple of showstoppers and am not really sure where to go from here. I've raised bugs for both of them (#17680, #17692) and had no response so far - have I perhaps overestimated how much chan_skinny is in use these days, or do I need to follow another route? Unfortunately the developer who was looking after that channel driver (community developer) has been pulled off onto other projects it seems, so currently there isn't much support for chan_skinny. If your timeframe is just a week or so, you're probably going to be further ahead with chan_sip. Leif Madsen. -- _ -- 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] Register Attacks End of ENUM ?
On 10-07-25 11:50 AM, Administrator TOOTAI wrote: Le 25/07/2010 02:11, Norbert Zawodsky a écrit : Hello again! Hi after it being relatively quiet her for the last weeks, my Astrerisk server was the target of 3 of that nasty REGISTER attacks during the last days. [...] Do like most of us are acting: use fail2ban. That's pretty much the solution to that problem right there: fail2ban. Leif Madsen. -- _ -- 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] URgent - capturing 'answered'
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a $
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
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Re: [asterisk-users] URgent - capturing 'answered'
Zarko Zivanovic wrote:
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
You need to change *ALL* the # to $
Doug
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[asterisk-users] Problem with Zap-Sip calls.
The problem we are having with Asterisk is when we initiate a call via a Zap line and it goes out on a Sip line. When it goes out via Sip we hear no sound until the party we are calling answers the line. If the call were to go out Sip-Sip or Zap-Zap it works perfectly fine. It is only with the Zap-Sip calls. If anyone knows anything that could possibly help it would be greatly appreciated. I have checked many different things already and tried comparing Zap-Zap and Zap-Sip call logs. Thanks! -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] PBX Lua with Asterisk ODBC
Ahh ok, so I am only able to access the application/functions that are
available to the dialplan.
I was wondering if it would be possible to access the handle of the odbc
connection directly from the lua dialplan.
On 26/07/10 17:10, Leif Madsen wrote:
On 10-07-26 10:34 AM, Faisal Hanif wrote:
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
So in the dialplan, after you've modified func_odbc.conf you'd be able to do a
query like:
exten = start,1,Set(MyResult=${ODBC_GET_MY_DATA(banana_lollipop)})
How you structure that in pbx_lua I'm not sure, but you create the functions
with func_odbc.conf, which is probably the piece you're missing.
Leif Madsen.
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Re: [asterisk-users] URgent - capturing 'answered'
Hi! Depending on the version of Asterisk you are running you can call a macro or an agi as option to dial. These will be called when the line is answered and you can find the channel name of who answered. Do as he says, look at the M option to Dial. Philipp -- _ -- 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] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a $
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] URgent - capturing 'answered'
On 26 July 2010 17:27, Zarko Zivanovic outlaw...@gmail.com wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a $
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
The # prefix is probably something to do with the combination of
Ruby, MySQL and AGI - I am not particularly familiar with Ruby.
You do not say at what stage when you call your AGI in the call path,
and at what stage you access the database.
1) Use AGI In order to make the call, and do not exit AGI until after
the call is over:
2) Call AGI after call is over
In case 1) you'll need the following somewhere in your script after
the call is answered or completed.
$agi.execute('Set(LOC=${CDR(dstchannel)})')
In case 2) just put
Set(LOC=${CDR(dstchannel)})
in your dialplan before calling the AGI
Then in your script try:
loc = $agi.get_variable('LOC')
CDR() is a built-in function rather than a variable, hence the need
for the indirection.
I am still guessing a bit here... Good luck.
Cheers,
Steve
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Re: [asterisk-users] URgent - capturing 'answered'
return $my.insert_id
else
$my.select_db('tvr2')
loc = $agi.get_variable('LOC')
$my.query(UPDATE call_log SET local = #{loc},
endtime = NOW() WHERE id = #{call_log_id})
raise 'Ne mogu da se ispisem iz call_log-a:
call_log_id = #{call_log_id}' if $my.affected_rows() != 1
end
nil
end
end
Maybe you figure out something.
Zarko
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: Monday, July 26, 2010 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 26 July 2010 17:27, Zarko Zivanovic outlaw...@gmail.com wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
The # prefix is probably something to do with the combination of
Ruby, MySQL and AGI - I am not particularly familiar with Ruby.
You do not say at what stage when you call your AGI in the call path,
and at what stage you access the database.
1) Use AGI In order to make the call, and do not exit AGI until after
the call is over:
2) Call AGI after call is over
In case 1) you'll need the following somewhere in your script after
the call is answered or completed.
$agi.execute('Set(LOC=${CDR(dstchannel)})')
In case 2) just put
Set(LOC=${CDR(dstchannel)})
in your dialplan before calling the AGI
Then in your script try:
loc = $agi.get_variable('LOC')
CDR() is a built-in function rather than a variable, hence the need
for the indirection.
I am still guessing a bit here... Good luck.
Cheers,
Steve
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Re: [asterisk-users] Problem with Zap-Sip calls.
The problem we are having with Asterisk is when we initiate a call via a Zap line and it goes out on a Sip line. When it goes out via Sip we hear no sound until the party we are calling answers the line. Search for progress and/or progressinband. -- _ -- 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] chan_skinny still maintained?
On 26 July 2010 17:17, Leif Madsen leif.mad...@asteriskdocs.org wrote: Unfortunately the developer who was looking after that channel driver (community developer) has been pulled off onto other projects it seems, so currently there isn't much support for chan_skinny. If your timeframe is just a week or so, you're probably going to be further ahead with chan_sip. Thanks Leif - much appreciated! I'm still happy to help as much as I can, and in fact would love to if I can.. In a week or so I'll need to actually use these phones, but will endeavour to set up a test server and try to keep at least one handset as skinny for testing purposes.. Cheers Jonathan -- If we knew what it was we were doing, it would not be called research, would it? - Albert Einstein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fail2ban - SuSEfirewall
I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything looks fine, except the machine restarts the firewall whenever the DHCP lease is renewed, thus flushing all the fail2ban rules (I think.). It seems to me that a quick fix would be to have the system restart fail2ban whenever the firewall is restarted. Has anyone else encountered this issue? .and come up with a solution? -- _ -- 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] URgent - capturing 'answered'
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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Re: [asterisk-users] URgent - capturing 'answered'
If all you need to do is the the channel name of the channel that answered the
phone why are you doing so much work? Version 1.4 allows for an agi to be
called when the dial command is answered. Version 1.6+ allows an agi as well as
a macro to be called. You can find the channel that answered a multi channel
dial command. Is this not what you wanted to know?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 10:40 AM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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http://www.eset.com
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Re: [asterisk-users] Management interface
On 26/07/10 13:15, Tony LaMear wrote: I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 *Tony * I've been looking at ZenOSS, which appears to have an asterisk zenpack as well. http://www.zenoss.com/ I've not used it as of yet though. -- _ -- 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] URgent - capturing 'answered'
On 7/26/2010 1:40 PM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
This is what I do with Perl AGI scripts and it works fine. You need to
figure out how to output to a text file with Ruby. I don't think the
'system' command would work with Ruby. Start with a basic AGI script
and test wether you can write to a file or not. That is the best way to
troubleshoot.
Andres
http://www.neuroredes.com
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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Re: [asterisk-users] URgent - capturing 'answered'
On Mon, 26 Jul 2010, Andres wrote: When I troubleshoot AGI scripts, I output stuff to text files for debugging purposes. I suggest you output all your variables to a file and then you will learn if the variables do have the info you need. Something like: $message=/bin/echo my variables are '$loc', '$variable1', '$variable2', etc /tmp/variables.txt; system($message); I prefer syslog(). ) You don't litter your system with little files. ) You get nicely timestamped messages you can centralize across servers. ) You can control how much verbosity you want by setting the logging priority. ) You can vary the logging priority at run time. ) You can leave the logging code in place in production. I code all of my AGIs to recognize (via getopt_long()) --debug and --verbose command line options. When something weird starts to happen, I can enable debugging in the dialplan and debug the code that is running in production. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URgent - capturing 'answered'
On Mon, 26 Jul 2010, Zarko Zivanovic wrote: I did try what you said, but it didnt create any files: $message=/bin/echo my variables are '$loc', '$variable1', '$variable2' /tmp/variables.txt; system($message); I'm just a c weenie, but that syntax would execute a command named $message, not the value of the variable $message. Would system($message); do what you want? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Management interface
I use a custom script that I run using SNMP, and graph that using cacti. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister Sent: Monday, July 26, 2010 13:57 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Management interface On 26/07/10 13:15, Tony LaMear wrote: I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 Tony I've been looking at ZenOSS, which appears to have an asterisk zenpack as well. http://www.zenoss.com/ I've not used it as of yet though. -- _ -- 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] Fail2ban - SuSEfirewall
On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrenga li...@torrenga.com wrote: I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything looks fine, except the machine restarts the firewall whenever the DHCP lease is renewed, thus flushing all the fail2ban rules (I think…). It seems to me that a quick fix would be to have the system restart fail2ban whenever the firewall is restarted. Has anyone else encountered this issue? …and come up with a solution? I believe there's a way to make the rules persist in a file. (see the fail2ban docs) /r -- _ -- 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] Fail2ban - SuSEfirewall
Randy R wrote: On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrengali...@torrenga.com wrote: I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything looks fine, except the machine restarts the firewall whenever the DHCP lease is renewed, thus flushing all the fail2ban rules (I think…). It seems to me that a quick fix would be to have the system restart fail2ban whenever the firewall is restarted. Has anyone else encountered this issue? …and come up with a solution? I believe there's a way to make the rules persist in a file. (see the fail2ban docs) /r Why isn't the Asterisk box on a static IP on the LAN? That seems to be asking for trouble using DHCP. John Novack -- Dog is my Co-pilot -- _ -- 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] PBX Lua with Asterisk ODBC
On Mon, 2010-07-26 at 12:10 -0400, Leif Madsen wrote:
On 10-07-26 10:34 AM, Faisal Hanif wrote:
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
So in the dialplan, after you've modified func_odbc.conf you'd be able to do
a
query like:
exten = start,1,Set(MyResult=${ODBC_GET_MY_DATA(banana_lollipop)})
With pbx_lua this would be something like:
MyResult = channel.ODBC_GET_MY_DATA(banana_lollipop):get()
How you structure that in pbx_lua I'm not sure, but you create the
functions
with func_odbc.conf, which is probably the piece you're missing.
Leif Madsen.
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Re: [asterisk-users] PBX Lua with Asterisk ODBC
On Mon, 2010-07-26 at 17:43 +0100, Bruce McAlister wrote: Ahh ok, so I am only able to access the application/functions that are available to the dialplan. I was wondering if it would be possible to access the handle of the odbc connection directly from the lua dialplan. Currently there is no way to directly access the odbc connection handle directly from pbx_lua. You can use a library like LuaSQL to connect directly to a database from pbx_lua, but you are probably be better off using asterisk's ODBC tools via func_odbc. That is the closest you can get to directly accessing the connection handle. -- Matthew Nicholson Digium, Inc. | Software Developer -- _ -- 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] Fail2ban - SuSEfirewall
On Mon, Jul 26, 2010 at 12:19 PM, John Novack jnov...@stromberg-carlson.org wrote: Why isn't the Asterisk box on a static IP on the LAN? That seems to be asking for trouble using DHCP. I was assuming he meant the ISP DHCP renewal. /r -- _ -- 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] Fail2ban - SuSEfirewall
On Monday 26 July 2010 14:19:58 John Novack wrote: Randy R wrote: On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrengali...@torrenga.com wrote: I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything looks fine, except the machine restarts the firewall whenever the DHCP lease is renewed, thus flushing all the fail2ban rules (I think…). It seems to me that a quick fix would be to have the system restart fail2ban whenever the firewall is restarted. Has anyone else encountered this issue? …and come up with a solution? I believe there's a way to make the rules persist in a file. (see the fail2ban docs) /r Why isn't the Asterisk box on a static IP on the LAN? That seems to be asking for trouble using DHCP. If the LAN is using an RFC-compliant DHCP server (read: not Microsoft), then it makes utterly no difference; as long as the machine is up whenever its lease expires and not too many MAC addresses are on the LAN, then it will always get exactly the same IP. The problem sounds like fail2ban is failing to write the new rules to a permanent file, which would otherwise allow the rules to persist after a reboot. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe
Hi guys,
i'm trying to use the featuremap of features.conf inside the app meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hangup();
};
in features.conf:
toca_macaco = 123, peer, Playback,tt-monkeys
But, if, inside the room, I press *123* the sound file tt-monkeys it's not
executed.
If a put DYNAMIC_FEATURES in a Dial string it works..
Any idea of how to use dtmf in the meetme app???
Thanx!!
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Re: [asterisk-users] MeetMe
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe Figueiredo Subject: [asterisk-users] MeetMe toca_macaco = 123, peer, Playback,tt-monkeys But, if, inside the room, I press 123 the sound file tt-monkeys it's not executed. Thanx!! As I recall, the DTMF feature in Meetme is Single-Digit; therefore if you change 123 to 9 (or some other single digit), you should get the desired result. (change 123 to 9 in features.conf). -- _ -- 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] MeetMe
Danny, didn't work... I didn't find other option to make meetme accpet dtmf but F. On Mon, Jul 26, 2010 at 5:25 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felipe Figueiredo *Subject:* [asterisk-users] MeetMe toca_macaco = 123, peer, Playback,tt-monkeys But, if, inside the room, I press *123* the sound file tt-monkeys it's not executed. Thanx!! As I recall, the DTMF feature in Meetme is Single-Digit; therefore if you change 123 to 9 (or some other single digit), you should get the desired result. (change 123 to 9 in features.conf). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe
I think there is a mis-communication here; If you changed features.conf so that toca_maccao = 123 . is now toca_maccao = 9, then if you press 9, monkeys should play. -- _ -- 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] Fail2ban - SuSEfirewall
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Monday, July 26, 2010 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fail2ban - SuSEfirewall Randy R wrote: On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrengali...@torrenga.com wrote: Why isn't the Asterisk box on a static IP on the LAN? That seems to be asking for trouble using DHCP. Not really; the trick is to assign an fixed IP address to the Mac address (with a host statement in ISC DHCP, or a reservation in Windows DHCP). I am a big fan of centralized management, so I prefer to do that rather than have static IP addresses on the network (except of course where absolutely essential). For the OP: maybe a workaround is to assign a fixed IP address from your DHCP server and use a very long lease time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VPMADT032 Failed! Unable to ping the DSP (2)!
Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as of a week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules and 1 VPMADT032 Module, hooked up to 5 analog lines. I get the error message referenced in the subject in my dmesg output everytime I load / reload DAHDI using the command system dahdi start/restart. When I make an outbound call over these analog lines, I'm told in the CLI WARNING[22601]: chan_dahdi.c:2685 dahdi_enable_ec: Unable to enable echo cancellation on channel 5 (No such device), and there is noticeable echo on the line, hearable by the caller inside the office, not by the recipient outside the office. When I make an inbound call, I get the same message, with echo still heard by the person inside the office, and not outside. I've run fxotune -i 4 -e 5 also, and still receive the same output. Below is my chan_dahdi.conf, my system.conf, my dmesg output while loading dahdi, if you need anything else let me know: *** chan_dahdi.conf *** [trunkgroups] [channels] usecallerid = yes hidecallerid = no threewaycalling = no echocancel = yes echocancelwhenbridged = no rxgain = 0.0 txgain = 0.0 group = 1 echocancel = yes signalling = fxs_ks context = incoming channel = 1-5 *** system.conf *** # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/0 Wildcard AEX800 Board 1 (MASTER) fxsks=1-5 # Global data loadzone= us defaultzone = us *** dmesg output *** dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.3.0.1 PCI: Enabling device :03:08.0 ( - 0003) ACPI: PCI Interrupt :03:08.0[A] - GSI 19 (level, low) - IRQ 177 wctdm24xxp :03:08.0: Port 1: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: Port 2: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: Port 3: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: Port 4: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: Port 5: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: Port 6: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: Port 7: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: Port 8: Installed -- AUTO FXO (FCC mode) wctdm24xxp :03:08.0: VPM100: Not Present wctdm24xxp :03:08.0: Booting VPMADT032 wctdm24xxp :03:08.0: VPMADT032 Failed! Unable to ping the DSP (2)! wctdm24xxp :03:08.0: Found a Wildcard TDM: Wildcard AEX800 (0 digital modules, 8 analog modules) dahdi: Registered tone zone 0 (United States / North America) wctdm24xxp :03:08.0: -- Setting echo registers: wctdm24xxp :03:08.0: -- Set echo registers successfully The last two lines of the dmesg output then repeat over and over again over time in the dmesg output. Any help you can offer would be much appreciated! -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_skinny still maintained?
Jonathan wrote: I've managed to acquire a few Cisco handsets (7905, 7920) and would like to use them with Asterisk. Rather than simply switching to the SIP firmware I thought I'd use these with chan_skinny - partly because this is Cisco's primary firmware and therefore the phones might be more stable, and partly to help test chan_skinny as it seems to be generally underused. (Is functionality identical across both firmwares?) However, I've come across a couple of showstoppers and am not really sure where to go from here. I've raised bugs for both of them (#17680, #17692) and had no response so far - have I perhaps overestimated how much chan_skinny is in use these days, or do I need to follow another route? The problem in 17680 has been worked on a couple of times and I believe the issue is not actually in chan_skinny, although it seems easiest to trigger from that channel. I had thought that the problem described in 17692 had also been put to rest, but the more I think about it, I seem to remember a potential fix was deferred pending a re-write of the subchannel handling code. I'll dig around in my archives to see if I can find my old patches for either of these. I'm not an Asterisk developer but am happy to spend some time this week resolving the problems. Unfortunately I need the phones next week, so may have to end up taking the defeatist approach of switching to the SIP firmware :( Regardless of whether the fixes were available, there is no way they would be reviewed, merged and released within the next week... Dan -- _ -- 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] VPMADT032 Failed! Unable to ping the DSP (2)!
On 07/26/2010 05:48 PM, Warren Selby wrote: Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as of a week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules and 1 VPMADT032 Module, hooked up to 5 analog lines. I get the error message referenced in the subject in my dmesg output everytime I load / reload DAHDI using the command system dahdi start/restart. When I make an outbound call over these analog lines, I'm told in the CLI WARNING[22601]: chan_dahdi.c:2685 dahdi_enable_ec: Unable to enable echo cancellation on channel 5 (No such device), and there is noticeable echo on the line, hearable by the caller inside the office, not by the recipient outside the office. When I make an inbound call, I get the same message, with echo still heard by the person inside the office, and not outside. I've run fxotune -i 4 -e 5 also, and still receive the same output. You should contact Digium technical support for assistance with this. They can walk you through the troubleshooting process. The most immediate thing you can do to limp along is enable sw echocan by editing /etc/modprobe.d/dahdi and add options wctdm24xxp vpmsupport=0 and ensure you have software echocan configured in /etc/dahdi/system.conf, which dahdi_genconf will do by default. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan Singh Jandu wrote: Excellent! I finally got it working, it was ODBC drivers issue actually. Installed the proper compatible version and its working. I thought that might be the case. There are still few errors which i see on asterisk console: [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: Realtime table book...@meetme requires column 'members', but that column does not exist! WMM does not use that column. You can disable it by Setting logmembercount=no in meetme.conf Also when i try to click the conference to manage it realtime it gives me Error connection to the manager! Following are the database files which i used: /web-meetme/cbmysql/db-admin-user-create.txt /web-meetme/cbmysql/db-table-create-v6.txt /web-meetme/cbmysql/db-tables-v6.txt Am i missing something here now? The WMM web interface used the Asterisk manager interface to monitor and manage conferences. The readme file documents the required changes to manager.conf. Sorry for the delay responding, I was on vacation last week with no email access. Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan Singh Jandu wrote: OK, now i added the column members in the table booking manually. and disabled selinux to have this working. Now i am struggling with recording option in webmeetme. Not sure on how to enable it, though m checking the checkbox while creating the conference. But where does this save and how to retrieve it? The location of the recordings is set in lib/defines.php as RECORDING_PATH, which defaults to /var/lib/asterisk/sounds/conf-recordings/ You can listen to the recordings after the conferences scheduled stop time by looking at the Past conferences page and clicking on the speaker icon next to the conference number. A couple of items to note- 1. You may have to check the path to ensure it exists and that the asterisk process can write to it. 2. Your web service accounts needs read permissions for that path 3. The speaker icon only displays if a recording exists. Dan -- _ -- 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] MeetMe
That was exactly what I did...but didn't work if I insert the option p in the meetme on the dialplan, I can leave the room pressing #, so dtmf is working fine On Mon, Jul 26, 2010 at 6:07 PM, Danny Nicholas da...@debsinc.com wrote: I think there is a mis-communication here; If you changed features.conf so that toca_maccao = 123 … is now toca_maccao = 9, then if you press 9, monkeys should play. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_skinny still maintained?
Hi Dan, On 26 July 2010 23:50, Dan Austin dan_aus...@phoenix.com wrote: I'll dig around in my archives to see if I can find my old patches for either of these. Many thanks - I'm happy to test patches if I can do so. At least I can contribute in that way, even if I'm not directly contributing wonderful code modules.. Regardless of whether the fixes were available, there is no way they would be reviewed, merged and released within the next week... I can hope, right? :-) Cheers, Jonathan -- If we knew what it was we were doing, it would not be called research, would it? - Albert Einstein -- _ -- 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] chan_skinny still maintained?
Jonathan wrote: On 26 July 2010 23:50, Dan Austin dan_aus...@phoenix.com wrote: I'll dig around in my archives to see if I can find my old patches for either of these. Many thanks - I'm happy to test patches if I can do so. At least I can contribute in that way, even if I'm not directly contributing wonderful code modules.. Regardless of whether the fixes were available, there is no way they would be reviewed, merged and released within the next week... I can hope, right? :-) The transfer issue is straight forward with two problems- 1. We always assume we have a second sub-channel (we don't, and When we don't we should create one) 2. We forget to tell the phone we have gone back on hook. The first is 16 lines copied from the redial softkey and the second is a simple callstate update. Very limited testing, but a patch will be on the bugtracker soon. The park issue is indeed very familiar, but I do not see any patches in my archive. The system is trying to playback the parked extension, but the parking channel has already been masqueraded away. I have a hack patch for that, but someone who understands the park/features code may have a better fix. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring X-lite for a remote user
I have asterisk running at home, a friend would be traveling out of the country and I want him to be able to put a call through from his remote location, I am wondering how I would configure the X-lite client on his pc so he would be able to call through assuming my public address is A.B.C.D and the static address the asterisk machine is on is 192.168.0.3. Thanks in anticipation _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] Configuring X-lite for a remote user
Configuring x-lite is a smaller problem here, do you have on your router your public IP ported to private IP at all and have you tested it before? As for x-lite check it on my website at http://visionvoip.com/help/x-lite.php Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-26 8:51 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: I have asterisk running at home, a friend would be traveling out of the country and I want him to be able to put a call through from his remote location, I am wondering how I would configure the X-lite client on his pc so he would be able to call through assuming my public address is A.B.C.D and the static address the asterisk machine is on is 192.168.0.3. Thanks in anticipation -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring X-lite for a remote user
To have your asterisk box reachable from internet you must configure static nat on your router to get sip traffic to the public Ip redirected to your internal ip. Make sure that sip and rtp traffic are not bloked by firewall. And configure xlite to connect to your public ip address. Adolphe Cher-aime From my Iphone On Jul 26, 2010, at 7:48 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: I have asterisk running at home, a friend would be traveling out of the country and I want him to be able to put a call through from his remote location, I am wondering how I would configure the X-lite client on his pc so he would be able to call through assuming my public address is A.B.C.D and the static address the asterisk machine is on is 192.168.0.3. Thanks in anticipation Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Si gn up now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring X-lite for a remote user
On Mon, Jul 26, 2010 at 6:14 PM, Adolphe Cher-aime achera...@gmail.com wrote: To have your asterisk box reachable from internet you must configure static nat on your router to get sip traffic to the public Ip redirected to your internal ip. Make sure that sip and rtp traffic are not bloked by firewall. And configure xlite to connect to your public ip address. Adolphe Cher-aime From my Iphone On Jul 26, 2010, at 7:48 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: I have asterisk running at home, a friend would be traveling out of the country and I want him to be able to put a call through from his remote location, I am wondering how I would configure the X-lite client on his pc so he would be able to call through assuming my public address is A.B.C.D and the static address the asterisk machine is on is 192.168.0.3. Thanks in anticipation Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dynamic hostnames are pretty useful too -- _ -- 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] MeetMe
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
Hi guys,
i'm trying to use the featuremap of features.conf inside the app meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hangup();
};
in features.conf:
toca_macaco = 123, peer, Playback,tt-monkeys
1) There is no peer when you invoke MeetMe. There is only a single call leg.
You therefore want self or caller.
2) Kill the spaces on this line. All of them. Note that self, caller,
or peer do not match anything and will thus signal Invalid 'ActivateOn'
specification for feature... at boot or reload. Similarly, there is a
dialplan application named Playback, but there is no dialplan application
named Playback.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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[asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Hi Guys, I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. The Asterisk Queue Analyzer is to serve as the graphic tool for call center or pbx admins. It will pull the info in queue.log and in MySQL asterisk CDR to create a graphic bar or to report on each extension that received the queue calls, etc... The Asterisk Log Analyzer is to analyze the log and to show any serious errors or as bonus maybe send out e-mails to admin and to e-mail any downtime during the day. Please note that I am not talking about making my own scripts to analyze and output this data as I know it exists in the system but rather am looking for a project that has done it already. Thanks, Bruce -- _ -- 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] Problem with Zap-Sip calls.
You may need to add r as option perameter to dial command. Regards, Faisal Hanif On 7/26/2010 9:39 PM, Chris Ramirez wrote: The problem we are having with Asterisk is when we initiate a call via a Zap line and it goes out on a Sip line. When it goes out via Sip we hear no sound until the party we are calling answers the line. If the call were to go out Sip-Sip or Zap-Zap it works perfectly fine. It is only with the Zap-Sip calls. If anyone knows anything that could possibly help it would be greatly appreciated. I have checked many different things already and tried comparing Zap-Zap and Zap-Sip call logs. Thanks! -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crashes to start if compiled with pbx_lua on latest updated CentOS
Hi, I have tried number of time if we update any CentOS system (or use latest CentOS version) then compile asterisk 1.6.2 with pbx_lua support, asterisk will crash on starting and will give a core dump. Issue is easy to produce, Install latest CentOS on a system. Install LUA LUA Headers using YUM. Download and Compile latest release of asterisk 1.6.2. Try to start start asterisk in console mode. It will crash on LUA and will give a core dump Did any one got it solved? If yes how? Regards, Faisal Hanif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] urgent:how to transfer a call using asterisk FAGI
Hi, I have xlite registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==1500,1,AGI(localhost//hello.agi So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard as this is very urgent. Thanks Regards, Jahnavi. -- _ -- 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
