Re: [asterisk-users] LUA
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Taylor Sent: Thursday, July 18, 2013 11:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] LUA On 07/18/2013 08:56 AM, jacob.e.mi...@l-3com.com wrote: I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the make linux install command. I can execute lua scripts via the command line, but asterisk configure script is unable to find the installation of Lua. I am on a closed network, so no access to the internet so I am not able to just install Lua using yum. OS CentOS 6.4 Asterisk version 1.8.13.0 11.4 $ find / -name *lua* /usr/local/include/lua.h /usr/local/include/lua.hpp /usr/local/include/lualib.h /usr/local/include/luaconf.h /usr/local/lib/lua /usr/local/lib/liblua.a /usr/local/bin/luac /usr/local/bin/lua /usr/lib64/liblua-5.1.so /usr/bin/luac /usr/bin/lua You don't mention it here, so I have to ask if you tried using --with-lua=/usr/local as an argument to configure. -- Dan Sorry it has taken me so long to get back to this, but I have tried the -with-lua=/usr/local but I would get an error during the configure script, something about LUA being broken or not present. I was able to get approval to download the rpms for lua lua-devel and move them to the system using a USB drive. After running the make linux uninstall on the source installation, and then install both RPMs for lua everything is work correctly now. Thanks for all the help, I would still like to see the configure script work correctly on a source install rather than just with the RPMs. Jacob -- _ -- 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] LUA
I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the make linux install command. I can execute lua scripts via the command line, but asterisk configure script is unable to find the installation of Lua. I am on a closed network, so no access to the internet so I am not able to just install Lua using yum. OS CentOS 6.4 Asterisk version 1.8.13.0 11.4 $ find / -name *lua* /usr/local/include/lua.h /usr/local/include/lua.hpp /usr/local/include/lualib.h /usr/local/include/luaconf.h /usr/local/lib/lua /usr/local/lib/liblua.a /usr/local/bin/luac /usr/local/bin/lua /usr/lib64/liblua-5.1.so /usr/bin/luac /usr/bin/lua Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422 image001.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
Re: [asterisk-users] LUA
I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org http://www.lua.org/ site, and I have installed via the “make linux install” command. I can execute lua scripts via the command line, but asterisk configure script is unable to find the installation of Lua. I am on a closed network, so no access to the internet so I am not able to just install Lua using yum. you're kidding right ? Why not just plug in the box somewhere else, do your install and move it back ? OS CentOS 6.4 Asterisk version 1.8.13.0 11.4 $ find / -name *lua* /usr/local/include/lua.h /usr/local/include/lua.hpp /usr/local/include/lualib.h /usr/local/include/luaconf.h /usr/local/lib/lua /usr/local/lib/liblua.a /usr/local/bin/luac /usr/local/bin/lua /usr/lib64/liblua-5.1.so /usr/bin/luac /usr/bin/lua Jacob Miles Software Engineer jacob.e.mi...@l-3com.com mailto:jacob.e.mi...@l-3com.com 903.457.4422 While a valid question, I have worked with clients on a closed military base where temporarily moving a box that has been secured back to an unsecured network would get you thrown off base and most likely result in criminal charges being filed. Not saying that is what Jacob is up against, but there are reasons that once a box is put somewhere you can't just move it back. This is very close to why I am unable to move the box to an open network connection. As well I do not have access to create/install my own yum repo, I am stuck using the box as it is. Is there a reason why the configure script does not find Lua in its default installation directory? Yes, I know that some distros package managers will install packages into difference locations based on how the install package was configured and created. But as a default I would think the configure script should look for items where the actual developer of product places them via their install script. What steps would I need to do to get the configure script to look in the correct location for a default installation of Lua from source. Or what location is the configure script looking for the Lua file to be in and I can manually move them to where asterisk is looking. Maybe the files are in the correct location but are named differently than what the script is looking for? Jacob image001.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
Re: [asterisk-users] AMI timeouts
Are you using a 3rd party java library such as asterisk-java (https://github.com/srt/asterisk-java), or are you doing your own Java AMI connector? I use asterisk-java and it has been working great. Jacob -- _ -- 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] Cisco SSCP to SIP
Even with the Cisco SIP firmware on the phones you still have to provide the XML configuration files to the phone via the TFTP. You need to have the XMLDefault.cnf.xml and SEPMAC ADDRESS.cnf.xml at the least... Jacob -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Courchesne Sent: Monday, June 17, 2013 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco SSCP to SIP Hi all, I'm trying to convers some Cisco SSCP phones to the SIP formware. The phone boots, I see it tries to fetch a bunch of files on my TFTP: Jun 17 09:37:45 firewall dnsmasq-dhcp[21202]: DHCPACK(eth2) 192.168.10.103 6c:50:4d:da:f0:67 SEP6C504DDAF067 Jun 17 09:38:10 firewall in.tftpd[22666]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22666]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22667]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22667]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22668]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:10 firewall in.tftpd[22668]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22669]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22669]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22671]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22671]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22672]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22672]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22673]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22673]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22674]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:12 firewall in.tftpd[22674]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22675]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22675]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22676]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22676]: sending NAK (1, File not found) to 192.168.10.103 But none of those are the SIP firmware filename I downloaded... Any hints ? 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] Cisco SSCP to SIP
This is for signed XML files, some of the newer models require signed files for security. Is there a reason to use SIP? There is a really good SCCP module for asterisk (chan-sccp-b http://sourceforge.net/projects/chan-sccp-b/ ). Usually you have to set in the SEPMAC ADDRESS.cnf.xml what firmware file to download, if it then finds the firmware file on the TFTP server it will upload and install the new firmware. This process can be difference depending on which Cisco phone you are using. When possible use Ciscos website for instructions on changing the firmware! Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Courchesne Sent: Monday, June 17, 2013 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco SSCP to SIP Yes I'm aware of the provisioning files, but first I need to have the freaking think update to an SIP firmware ;-) I found this how-to which is the best I found so far: http://www.adrianandgenese.com/blogger/2011/02/16/how-to-upgrade-or-conv ert-a-cisco-ip-79xx-7940-7960-794x-796x-797x-phone-to-sip-or-sccp/ However I see that the phone tries to fetch XMLDefault.cnf.xml.sgn and not XMLDefault.cnf.xml... Any idea what is the sgn extension ? Andre On 2013-06-17, at 9:48 AM, jacob.e.mi...@l-3com.com wrote: Even with the Cisco SIP firmware on the phones you still have to provide the XML configuration files to the phone via the TFTP. You need to have the XMLDefault.cnf.xml and SEPMAC ADDRESS.cnf.xml at the least... Jacob -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Courchesne Sent: Monday, June 17, 2013 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco SSCP to SIP Hi all, I'm trying to convers some Cisco SSCP phones to the SIP formware. The phone boots, I see it tries to fetch a bunch of files on my TFTP: Jun 17 09:37:45 firewall dnsmasq-dhcp[21202]: DHCPACK(eth2) 192.168.10.103 6c:50:4d:da:f0:67 SEP6C504DDAF067 Jun 17 09:38:10 firewall in.tftpd[22666]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22666]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22667]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22667]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22668]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:10 firewall in.tftpd[22668]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22669]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22669]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22671]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22671]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22672]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22672]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22673]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22673]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22674]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:12 firewall in.tftpd[22674]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22675]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22675]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22676]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22676]: sending NAK (1, File not found) to 192.168.10.103 But none of those are the SIP firmware filename I downloaded... Any hints ? 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 -- _ --
[asterisk-users] Skinny directmedia
Asterisk 11 CentOS 6.4 Cisco 7971 phones Does chan_skinny support directmedia? Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422 image001.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
Re: [asterisk-users] Joining an astablished call
The best way I have found to do this is to use ChanSpy/ExtenSpy and then use the wisper/barge modes. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Monday, May 06, 2013 9:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Joining an astablished call In the telephony world that is known as barge-in and is a programmable option granting that right to specific extension(s) in systems that normally have automatic privacy. Not all electronic key and hybrid systems have automatic privacy, though most do. John Novack neo haux wrote: Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu It's just a normal call between to channels that I have to join for few minutes. Regards -- _ -- 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 -- 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
[asterisk-users] Caller ID is not persisted when using Channel Redirect
Is there a work around for Caller ID information not being persisted when using the CLI or AMI Channel Redirect. A calls B (caller id is displayed), B transfers call to C (no caller id is displayed on phone c). Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422 image001.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
Re: [asterisk-users] extensions.conf / test DID
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Hostdnsmgr Username Refresh
StateReg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI
Here is the dial plan:
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please?I think something to do w/ incoming is incorrect.
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
As well doesn't the Goto need to closing )?
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Re: [asterisk-users] How to install in /usr/local/sbin instead of/usr/sbin ?
If this is the case then doing make install DESTDIR=../local/sbin should install in the /usr/local/sbin directory. It looks to be using a relative path starting in /usr/sbin/ Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Tuesday, February 12, 2013 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to install in /usr/local/sbin instead of/usr/sbin ? Hi, Reading comment in the bottom of https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Aster isk, I thought I could install asterisk 11 in non-standard locations such as /usr/local/sbin simply typing (from source directory): make install DESTDIR=/usr/local/sbin Doing so seems to install elsewhere For instance, make install DESTDIR=/usr/sbin installs runtime asterisk in /usr/sbin/usr/sbin directory. Am I correctly understanding the wiki page ? What is the appropriate command ? Regards -- _ -- 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] How to get louder voice ?
Not sure if this is what you want but you can always set the TX and RX Gain values via the dialplan. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Tuesday, November 13, 2012 4:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to get louder voice ? Hello, I have the following case. A customer is a heavy Meetme/audio conference user. He is equiped with Polycom SS2W (DECT SoundStation 2W audio conference station). Users complain they often do not hear the other party loud enough. The setup is then: Remote party --- PSTN/ISDN--- Asterisk ---SIP--- Kirk300 ---DECT--- SS2W My questions are: 1. How can I measure audio strength/loudness/quality and strip social/psychological interferences off ? 2. Is there any builtin mechanism inside Asterisk (this setup is 1.6.1 but upgrade is possible) that can change call volume ? 3. Given my setup is purely digital, could it be the source of calls not being loud enough ? 4. Suggestions ? Comments Regards -- _ -- 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
