[Asterisk-Users] mangle + to 00
hi,
i want change prefix from +XXX. to 00XXX.
but this doesnt work
[incoming]
exten = _+.,1,SetCIDnum(00${CALLERID:1})
exten = _+.,2,goto(incoming,${EXTEN},1)
exten = _X.,1,Noop(CALLERID: ${CALLERID})
exten = _X.,2,goto(route,${EXTEN},1)
can you help?
---
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA - http://www.fpf.slu.cz
LCNA- http://lcna.slu.cz
===
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Re: [Asterisk-Users] chan_cornet
i agree... no H.323 support for endpoint (HG3530) but you still have h.323 (for the moment only version 2.0) support for ip-trunking (HG3550). So what if you have the following setup. [OPTIPOINT400_HFA]--[HIPAT4K][oh.323][ASTERISK]--[OPTIPOINT400_SIP]. how can i configure ip-trunking from HI4K to asterisk? any example h323 conf for asterisk? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] :: Success Case = Motorola 62802-51 as FXO device ::
Hello list , I´d like to report a success case with a modem based on chipset : Motorola 62802-51. It works fine , and zaptel identifies as a X100P ( not clone ) . Red Alarms can be identified . :) This doesn´t occurred on MD3200 ambient chipsets. can you send us more info? driver,versions,logs, audio experience (echo, delay, ...) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1x fxs + 1x fxo transfer
hi, i have 1 PSTN line and ip or analog phone i need get call(with phone ip or analog) from PSTN and transfer it(i.e. to sales) to the asterisk on corporate network pstn - gw - asterisk | phone can you recommend me some hardware (cheapest than PC+fxo card+asterisk)? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel vanilla kernel
hi, to digium maybe some individuals: do you plan add zaptel drivers to vanilla kernel? for users is this very good thing --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.4 and more ... + rpm spec
Will be good, if somebody could provide rpms for every release and also rpm's with static compiled chan_oh323 and Asterisk-oh323 modules asterisk.spec for 1.0.5 is in attachment put this file into /usr/src/redhat/SPECS asterisk-1.0.5.tar.gz to the /usr/src/redhat/SOURCES cd /usr/src/redhat/SPECS rpmbuild -ba asterisk.spec if this file will be contained directly in the tarball (like openvpn or other good software), then simply run rpmbuild -ta asterisk-1.0.5.tar.gz --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.4 and more ... + rpm spec
Will be good, if somebody could provide rpms for every release and
also rpm's with static compiled chan_oh323 and Asterisk-oh323 modules
asterisk.spec for 1.0.5 is in attachment
put this file into /usr/src/redhat/SPECS
asterisk-1.0.5.tar.gz to the /usr/src/redhat/SOURCES
cd /usr/src/redhat/SPECS
rpmbuild -ba asterisk.spec
if this file will be contained directly in the tarball (like openvpn
or other good software), then simply run
rpmbuild -ta asterisk-1.0.5.tar.gz
sorry, file is in attachment now
---
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA - http://www.fpf.slu.cz
LCNA- http://lcna.slu.cz
===
%define version 1.0.5
%define nameasterisk
%define release 1
%define group Applications/Internet
%define copyright GPL
%define uname %{name}
%define gname %{name}
summary : A complete PBX in software
name: %{name}
version : %{version}
release : %{release}
group : %{group}
copyright : %{copyright}
url : http://www.asterisk.org/
vendor : www.digium.com
provides: %{name}
buildroot : /var/tmp/%{name}-%{version}
source0: %{name}-%{version}.tar.gz
%description
Asterisk is an Open Source PBX and telephony toolkit. It is, in a
sense, middleware between Internet and telephony channels on the bottom,
and Internet and telephony applications at the top. For more information
on the project itself, please visit the Asterisk home page at:
http://www.asterisk.org
%package devel
Summary : Development libraries and headers for Asterisk PBX package
Group : %{group}
requires: %{name} = %{release}
%description devel
Development Libraries and headers for Asterisk PBX package
%package webvmail
summary : Asterisk Web Voicemail
group : %{group}
requires: %{name} = %{release}
%description webvmail
SUID ROOT Perl cgi script for web based Voicemail retrieval.
%prep
%setup -q
%build
## make asterisk
# Replace /var/run by /var/run/asterisk since we don't run as root
sed -i
s/ASTVARRUNDIR=\$(INSTALL_PREFIX)\/var\/run/ASTVARRUNDIR=\$\(INSTALL_PREFIX\)\/var\/run\/%{name}/g
Makefile
make
%install
rm -rf %{buildroot}
mkdir -p %{buildroot}/usr/include/linux
mkdir -p %{buildroot}%{_sysconfdir}/sysconfig
mkdir -p %{buildroot}/var/www/{html,cgi-bin/astcc-admin}
mkdir -p %{buildroot}/var/run/asterisk
## install asterisk
mkdir -p %{buildroot}/var/www/{cgi-bin,html/_asterisk}
make INSTALL_PREFIX=%{buildroot} install
make INSTALL_PREFIX=%{buildroot} samples
make DESTDIR=%{buildroot} webvmail
install -D -m 0755 contrib/init.d/rc.redhat.asterisk
%{buildroot}%{_initrddir}/%{name}
install -m 755 contrib/scripts/addmailbox %{buildroot}/%{_sbindir}
# Override wrong absolute links
rm -f %{buildroot}%{_localstatedir}/lib/%{name}/sounds/vm \
ln -sf ../../../spool/%{name}/vm \
%{buildroot}%{_localstatedir}/lib/%{name}/sounds/vm
rm -f %{buildroot}%{_localstatedir}/lib/%{name}/sounds/voicemail \
ln -sf ../../../spool/%{name}/voicemail \
%{buildroot}%{_localstatedir}/lib/%{name}/sounds/voicemail
rm -f %{buildroot}%{_localstatedir}/spool/%{name}/vm \
ln -sf voicemail/default \
%{buildroot}%{_localstatedir}/spool/%{name}/vm
# fix samples installation
pushd %{buildroot}/%{_sysconfdir}/%{name}
for i in `find . -type f`; do
sed s,%{buildroot},,g $i $i.fix
mv -f $i.fix $i
done
popd
%clean
rm -rf %{buildroot}
%pre
# Add the %{name} user
/usr/sbin/useradd -c Asterisk PBX -G tty -s /sbin/nologin -r \
-d %{_localstatedir}/lib/%{name} %{uname} 2/dev/null || :
%post
# Register the %{name} service
/sbin/chkconfig %{name} --add
/sbin/chkconfig %{name} on
%preun
if [ $1 -eq 0 ]; then
/sbin/service %{name} stop /dev/null 21
/sbin/chkconfig %{name} --del
fi
%files
%defattr(- ,%{uname},%{gname})
/etc/rc.d/init.d/asterisk
/usr/lib/asterisk/modules/*
/usr/sbin/addmailbox
/usr/sbin/asterisk
/usr/sbin/astgenkey
/usr/sbin/astman
/usr/sbin/safe_asterisk
/usr/share/man/man8/asterisk.8.gz
%dir %{_sysconfdir}/%{name}
%attr(- ,%{uname},%{gname}) %{_localstatedir}/lib/%{name}
%attr(750,%{uname},%{gname}) %{_localstatedir}/run/%{name}
%attr(750,%{uname},%{gname}) %dir %{_localstatedir}/log/%{name}
%attr(750,%{uname},%{gname}) %dir %{_localstatedir}/log/%{name}/cdr-csv
%attr(750,%{uname},%{gname}) %dir %{_localstatedir}/spool/%{name}
%attr(750,%{uname},%{gname}) %dir %{_localstatedir}/spool/%{name}/vm
%attr(750,%{uname},%{gname}) %dir %{_localstatedir}/spool/%{name}/voicemail
%attr(750,%{uname},%{gname}) %dir
%{_localstatedir}/spool/%{name}/voicemail/default
%attr(640,%{uname},%{gname})
%{_localstatedir}/spool/%{name}/voicemail/default/1234/*
%attr(640,%{uname},%{gname}) %config(noreplace) %{_sysconfdir}/%{name}/*.conf
%attr(640,%{uname
[Asterisk-Users] ogg vorbis
hi, what are the reasons why ogg player is not included in asterisk?(for onhold music) technical, political, no coders? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP jitter?
Eric Wieling wrote: joachim wrote: Yes, It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). So? That's what CVS-HEAD is there for. Adding in experimental patches willy-nilly, especially ones that have the potential to cause huge problems, confounds attempts to isolate bugs and test functionality. Mark does a pretty good job of keeping the HEAD version solid enough to use in production, as most of us running it on a daily basis can attest. What stops you from applying the patches to your own copy, and then playing with it to your heart's content--like the rest of us? It would work just like it had really been put into CVS-HEAD. less testers less bug reports for production use is stable version (asterisk doesnt have good roadmap and versioning :( ) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] click to dial extension number functionality ?
By any web-user (ms explorer) to be able to call from a web-page to a
certain number/extension connected to one specific asterisk.
maybe this php script help you (switch caller/called and modify Exten:)
--originate.php--
?php
# configuration
$astip=192.168.0.1;
$astmanager=test;
$astpassword=isbest;
$mancmd=;
$wrets=;
$tech=SIP;
?
br
form action=originate.php method=get
CALLERinput type=text name=caller size=8 maxlength=18
CALLEDinput type=text name=called size=30 maxlength=30
input type=submit value=submit
/form
br
?php
if ( ( isset($_GET['caller'] )) ( $_GET['caller'] != )
( isset($_GET['called'] )) ( $_GET['called'] != ) )
{
$called = $_GET['called'];
$caller = $_GET['caller'];
$socket = fsockopen($astip,5038, $errno, $errstr);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: $astmanager\r\n);
fputs($socket, Secret: $astpassword\r\n\r\n);
fputs($socket, Action: Originate\r\n);
fputs($socket, Channel: $tech/$caller\r\n);
fputs($socket, Context: $caller\r\n);
fputs($socket, Exten: $called\r\n);
fputs($socket, Priority: 1\r\n);
fputs($socket, Callerid: $caller\r\n\r\n);
fputs($socket, Action: Logoff\r\n\r\n);
while (!feof($socket)) {
$wrets .= fread($socket, 8192);
}
fclose($socket);
echo pre;
echo ASTERISKMANAGEREND
$wrets
ASTERISKMANAGEREND;
echo /pre;
}
?
---
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA - http://www.fpf.slu.cz
LCNA- http://lcna.slu.cz
===
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Re: [Asterisk-users] Asterisk 1.0.6
Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the source as well as on the following web page: http://dev.asteriskdocs.org If you will be attending Spring VON in San Jose, be sure to come by the Asterisk Pavilion and say hello! A number of the Asterisk developers, including myself, will be there talking to people about Asterisk. there is unofficial fast mirror in europe (md5 will be useful) ftp://ftp.ipex.cz/pub/mirrors/ftp.asterisk.org/pub/telephony/asterisk/ --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: PA1688 Chipset IP Phones ATA's
For those of you who may be interest IAX2 loads are now available for the standard builds... http://www.aredfox.com/edownloadsiax2.htm Just a word of caution... Remember to change the ports over to 4569 from whatever. And don't forget to grab the palmtool from http://www.aredfox.com/download/tools/PalmTool.zip My own testing of IAX2 with both a phone and an ATA is that IAX2 is working very well :-) i have problem with upgrade i have phone like this http://www.voip-info.org/wiki-Atcom tested firmware is http://www.aredfox.com/download/English/program/iax2/PA168S.zip from debug 192.168.1.100: PHONE 192.168.1.100: V1.37.008 192.168.1.100: Updating ... Please Wait 192.168.1.100: upgrade binary mismatch any help? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: PA1688 Chipset IP Phones ATA's
i have problem with upgrade i have phone like this http://www.voip-info.org/wiki-Atcom tested firmware is http://www.aredfox.com/download/English/program/iax2/PA168S.zip from debug 192.168.1.100: PHONE 192.168.1.100: V1.37.008 192.168.1.100: Updating ... Please Wait 192.168.1.100: upgrade binary mismatch any help? i'm found the problem in the settings debug - no check --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk1.2 to 1.4 g711a fax
hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is changed in 1.4 in g711 pass-through faxing? thanks marek ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk1.2 to 1.4 g711a fax
i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is changed in 1.4 in g711 pass-through faxing? thanks Jitterbuffer behavior, maybe? jbenable=yes or no has no effect BUT i'm discover that with clear DIAL command fax works but if i use AGI (like a2billing etc) then fax FAIL any ideas? Marek Cervenka ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk1.2 to 1.4 g711a fax
i have problem with pass-through faxing
with this scenario
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen
virtual) - linksys ATA
i can fax to fax2mail on hylafax
but after upgrade asterisk2 to 1.4 faxing is not working
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen
virtual) - linksys ATA
configuration is same
do you hava any idea what is changed in 1.4 in g711 pass-through faxing?
thanks
Jitterbuffer behavior, maybe?
jbenable=yes or no has no effect
BUT i'm discover that with clear DIAL command fax works but if i use AGI
(like a2billing etc) then fax FAIL
any ideas?
can you someone confirm that faxing with this simple AGI script is
working? (phpagi is from phpagi.sf.net)
#!/usr/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi-answer();
$dialstr = SIP/asterisk1/1|300|HgL(61:61000);
$myres = $agi-exec(DIAL $dialstr);
$agi-hangup();
?
thanks!
Marek Cervenka
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Re: [asterisk-users] Faxing through a PAP2
On Fri, 10 Aug 2007, Carlos Chavez wrote: I usually have good results when using a regular fax machine connected to a PAP2T on a small network. I have a customer that has this setup in several offices. Lately I have noticed that recent versions of Asterisk have worse results with this fax setup that onlder versions. I have 3 new installations where they have Asterisk 1.4.9 with a TDM800P card. It is almost impossible to send or receive faxes using the PAP2T. Other offices still have version 1.2 and they rarely have problems faxing. i have same problems with asterisk 1.4 too --- Marek Cervenka === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redundancy / Failover
Any one succeeded to make _Redundancy* / Failover with asterisk 1.4.9 on centos with kernel 2.6.9-55.EL. ***_ Can you please send me the documentation link on how to or write down how to . hint yum -y install heartbeat (on node1 and node2) configure heartbeat if you have configuration in mysql then set up master-to-master replication (- www.mysql.com) or generate ssh keys priodically copy /etc/asterisk and /var/lib/asterisk/astdb from master node to slave node (astdb is needed because of sip registrations) question1: do you someone know how to _easily_ find out which node is master? (heartbeat) - now i have custom script for this question2: it's possible read registration data from astdb from python/php (or it is possible write sip registrations to mysql/sqlite? i do not want realtime because of NAT issues) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax pass-through
DEBUG[28047] pbx.c: Function result is '46' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '54' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'pstn' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'SIP/46-62bb' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Zap/1-1' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Dial' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Zap/g1/54||tT' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:50:35' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:50:54' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:52:00' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '85' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '66' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'ANSWERED' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'DOCUMENTATION' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '(null)' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '1139871035.6' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '(null)' any ideas? --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax pass-through
to state '0' (Unknown) Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '46' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '46' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '54' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'pstn' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'SIP/46-62bb' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Zap/1-1' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Dial' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Zap/g1/54||tT' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:50:35' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:50:54' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:52:00' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '85' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '66' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'ANSWERED' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'DOCUMENTATION' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '(null)' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '1139871035.6' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '(null)' any ideas? --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_ss7 0.10
Thanks for the update. I have Sangoma A104D and wanted to use ss7 signalling. I came accross chan_ss7 but found sifira is not in active development. But is this chan_ss7 stable and can be used in production server implementation. We are going to have 2 to 3 boxes with ss7 signalling using sangoma but I am looking for open source ss7 implementation which is chan_ss7. so need to know about stability and recommendation for using on production server. long term supported solution is libss7 from digium. but this depends on asterisk 1.6 which is not officialy stable chan_ss7 is now developed by www.dicea.dk. http://www.dicea.dk/company/downloads it's used on production servers. it is very stable solution --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming call popup
hi, can you recommend cleansimplestable solution for incoming call popup (in browser)? i'm using flash operator panel now but i want something without flash (maybe something in AJAX?) thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (announce) asterisk T.38 gateway
hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (announce) asterisk T.38 gateway
marek cervenka wrote: hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz And you will, of course, pass on 99% of the money to those who did 99% of the work, won't you? :-) if you want, it's no problem (sponsors please CC: steveu at coppice.org) but by now nobody respond ... (surprisingly) contract for primary developer(dafe) is exhausted. that's the reason for bounty request SpanDSP is good piece of software and BIG credit goes to you Steve thanks! --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk stops sending qualify
hi, i have problem with asterisk 1.4.20.1 (kernel 2.6.25.10, centos5, ztdummy+hrtimers) after some random time, asterisk stops sending qualify (monitored by wireshark) to peer (phone) before i'll go to bugs.digium.com is there someone with similar problem? thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 capable soft phone?
Does anyone know where I might purchase a G.722 capable SIP soft phone? Counterpath say that they offer one, but only in the OEM versions do they support G.722. I need only a couple of licenses. www.qutecom.org --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] SRTP implementation
Olle E Johansson wrote: 23 apr 2007 kl. 19.55 skrev Russell Bryant: John Todd wrote: To morph this into a -dev thread: if this patch were to become (again) useful and error-free, is there any objection or usefulness in adding it to TRUNK? Personally, I think there is, if there is a method by which SRTP can be activated or de-activated from within the dialplan based on prior shared secrets. However, I have heard others disagree and object that without signalling-based secure key exchange, SRTP is not worth the effort. Opinions? I agree with you. I think that is a reasonable approach. I can't speak for the quality of the patch itself as I have not reviewed it. But, if it works, I would guess that it would not be too bad to get it into trunk. Kevin and I earlier decided that we wanted to delay this until we had a complete security solution, with signalling based secure key exchange ;-) /O I have uploaded a new patch. This patch and also the previous supports MIKEY as well as sdescriptions. The MIKEY key management scheme uses transport encryption for transporting the keys securely over unsecured transports such as unencrypted SDP. There are several MIKEY flavors: Pre shared, DH-SIGN, RSA, RSA-R and DH-HMAC. The patch currently uses DH-HMAC for outgoing connections, using secret from sip.conf as the shared secret. http://www.voip-info.org/wiki/view/Asterisk+SRTP updated test srtp server (asterisk SVN-trunk-r61760 + latest SRTP patch) voice2.fpf.slu.cz test sip accounts 700:700 701:701 702:702 extensions.conf exten = 600,1,Set(_SIPSRTP=optional) exten = 600,n,Set(_SIPSRTP_CRYPTO=enable) exten = 600,n,Playback(demo-echotest) ; Let them know what's going on exten = 600,n,Echo ; Do the echo test exten = 600,n,Playback(demo-echodone) ; Let them know it's over exten = 600,n,hangup exten = 610,1,Set(_SIPSRTP=require) exten = 610,n,Set(_SIPSRTP_MIKEY=enable) exten = 610,n,Playback(demo-echotest) ; Let them know what's going on exten = 610,n,Echo ; Do the echo test exten = 610,n,Playback(demo-echodone) ; Let them know it's over exten = 610,n,hangup p.s. sorry for cross post --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.2.17 - 1.2.18 asterisk crash
hi, i am updated to latest asterisk stable (because of security problems), but now asterisk crashes within a hour log is clear do you someone have this problem too? --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Well an upgrade to 1.2.17 now results in blips in the audio, instead of it dropping. Guess it's time to go to SuperMicro. 1.2.17 ? (1.2.13 zaptel?) i have supermicro mobo(P8SCT) and have same problem with shared interrupts bash#lspci -bv | grep -i IRQ 5 --before-context=2 00:02.0 VGA compatible controller: Intel Corporation E7221 Integrated Graphics Controller (rev 05) (prog-if 00 [VGA]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, fast devsel, latency 0, IRQ 5 -- 00:1d.3 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #4 (rev 03) (prog-if 00 [UHCI]) Subsystem: Super Micro Computer Inc: Unknown device 7480 Flags: bus master, medium devsel, latency 0, IRQ 5 -- 02:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 -- 03:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Super Micro Computer Inc: Unknown device 02c6 Flags: bus master, fast devsel, latency 0, IRQ 5 can you someone explain what's mean by (zaptel 1.2.13 changelog) 2007-01-23 21:28 + [r1936] Matt Frederickson [EMAIL PROTECTED] * wcte11xp.c, wct1xxp.c, wctdm.c, wctdm24xxp.c: Make sure we don't clear the interrupt before we might have received it in shared interrupt line scenarios. --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRTP testers needed
please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRTP testers needed
On Fri, 2007-03-23 at 16:12 +0100, marek cervenka wrote: please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,gxp-2000, minisip, twikle, ...) Does this work on 1.2 or 1.4 too or is it trunk only? trunk only ... now no testers, no stable release --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?
I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? http://www.qutecom.org it is successor to openwengo --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRTP testers needed
please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,grandstream,aastra, qutecom, eyebeam, ...) digium need feedback for srtp inclusion to 1.6.3.0 http://bugs.digium.com/view.php?id=5413 if you need additional info, i'm on jabber - cerv...@njs.netlab.cz thanks! --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open source SIP client
can anybody help me to give Opensource SIP client information which can be modified as per our requirment http://www.qutecom.org --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] playing media(moh,prompts) from flash player
hi, i'm searching solution for playing media(moh,prompts,voicemail,recordings - wav format) from adobe flash player (web browser) flash cannot play wav directly (imho) i must convert files to any other format on-the-fly - i cannot use mp3 because of royalties - next option is swf (with ffmpeg), supported free audio codecs (http://en.wikipedia.org/wiki/Flash_Video#Format_details) * uncompressed PCM * ADPCM * AAC can you someone recommend solution/combination which works? tnx --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ekiga, Twinkle and from where to start with open source
Hi All; I am looking to start develop an Softphone that has messanger feature (voice and text, who is online also), anyone can advise for the best link to start with it, so they have open source for softphone that we can start on it from there? http://www.qutecom.org (platform - windows,linux,mac) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] looking for GXV-3000 users
hi, i want try Grandstream GXV-3000 video part. i'm looking for GXV users. i have asterisk-trunk available. please contact me privately (or at jabber:[EMAIL PROTECTED]) --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)
T38 passthrough doesn't seem to work in trunk at the moment. that's true http://bugs.digium.com/view.php?id=7679 http://bugs.digium.com/view.php?id=7844 t.38 in asterisk 1.4 http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] digium compatibility notes
hi, what is mean by partially incompatible in http://www.digium.com/en/docs/misc/compatibility_notes.php i have server with E7221+te110p mobo and i think i dont have any problems thanks --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: pocket pc + ilbc/g729
hi, can you recommend some pocket pc sip client with iLBC or G729? i'm googling over a hour and found nothing --- Marek Cervenka === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs order
hi, i have this topology pstn+(e1)asterisk1-asterisk2-sip client asterisk1,asterisk2 allow (g729,alaw) sip client prefer g729, then alaw can you someone describe codec negotiation when call for sip client arrive from pstn? (can i set g729 for calls from pstn? ) thanks --- Marek Cervenka === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ftp.digium.com HTTP mirror, Digium's FTP server
The ftp server has been broken for months. If you keep trying you will eventually get a listing or a file. i'm using ftp://ftp.ipex.cz/pub/mirrors/ftp.asterisk.org/ --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000 headset
Hi all What headset do people use with the GXP-2000? Any recommondations for or against particular models? i'm sent mail to [EMAIL PROTECTED], need info too btw i'm asked that will support IAX, they respond yes, if customers want it - write them --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gxp-2000 tftp cfg
hi, can you someone post tftp template for gxp-2000? like http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txt thanks --- Marek Cervenka === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip nat bug
hi, can you someone explain this bug? (or point me to number from bugs.digium.com) 2006-03-28 19:07 + [r15699] Olle Johansson [EMAIL PROTECTED] * channels/chan_sip.c: Fix breakage of NAT support for peers with qualify=yes. Thanks Damin for access to your system, sorry folks. thanks --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream GXV-3000
hi,
do you someone test this http://www.grandstream.com/y-gxv3000.htm?
video works? (it's have H264 video codec)
i want this topology
gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000
---
Marek Cervenka
LCNA- http://lcna.slu.cz
===
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[Asterisk-Users] Astricon - materials
hi, will be somewhere materials (videos, presentations) from astricon? thanks --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon - materials
marek cervenka wrote: hi, will be somewhere materials (videos, presentations) from astricon? Registered attendees will get information about the material soon. No videos where recorded this year. any chance for not registered? astricon was too far for me (europe) my english is terrible, but i can read if you have the materials, it's wrong to not use it (it can be for money) The 1.2 presentation I made together with Kevin has been available for a while at http://www.astricon.net/asterisk1-2/ and will be updated soon. nice intro to 1.2, thanks! --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sqlite + stable asterisk
hi, i have problem with compiling cdr_sqlite rhel4(gcc3.4.3) + sqlite3 (from fc4 - rebuilded) any ideas? gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-08/11/05-19:35:03\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC-c -o cdr_sqlite.o cdr_sqlite.c cdr_sqlite.c:38: error: syntax error before '*' token cdr_sqlite.c:38: warning: type defaults to `int' in declaration of `db' cdr_sqlite.c:38: warning: data definition has no type or storage class cdr_sqlite.c: In function `sqlite_log': cdr_sqlite.c:92: warning: implicit declaration of function `sqlite_exec_printf' cdr_sqlite.c: In function `unload_module': cdr_sqlite.c:153: warning: implicit declaration of function `sqlite_close' cdr_sqlite.c: In function `load_module': cdr_sqlite.c:166: warning: implicit declaration of function `sqlite_open' cdr_sqlite.c:166: warning: assignment makes pointer from integer without a cast cdr_sqlite.c:174: warning: implicit declaration of function `sqlite_exec' make[1]: *** [cdr_sqlite.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 --- Marek Cervenka === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] link quality monitor
hi, do you someone know tool that can get data like latency/bandwith/jitter/packet loss (in one program) - it must be functional behind nat - multiplatform (AJAX,java applet) - preferably on SIP and IAX ports - can be client/server - easy to use ;) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] show channels
hi, i have problem with showing actual channels asteriskshow chanels SIP/123456789-b6c4b2 [EMAIL PROTECTED] Up Busy() (last 2 chars are NOT showed) but the name of channel is longer asterisk show channel SIP/123456789-b6c4b290 how can i get full name of channel with asterisk -rqnx ? thanks --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip realtime
hi, i'm reading a lot docs about asterisk realtime but i cannot understand how works sip realtime static i need NAT/qualify for SIP. this is not possible with dynamic realtime i want - save data to sql - asterisk -rx reload to read config (sip.conf with sip users) from sql it is possible? can you point me to some examples? thanks --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t.38 asterisk-trunk
hi, can you anyone share experience with t.38 passthrough in asterisk trunk? (working configurations) do you have someone patton smartnode 4960? (t.38 + sipura/grandstream) http://www.patton.com/products/pe_products.asp?category=354MiDAS_SessionID=cff1cc1234ca4a928823912cba91343e and in this bug are questions without answers http://bugs.digium.com/view.php?id=5090 like can i have an ATA behind NAT? what mean chan_sip.c:4586 process_sdp: Unsupported SDP media type in offer: image 10912 udptl t38? --- Marek Cervenka FPF SLU OPAVA - http://www.fpf.slu.cz === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t.38 bounty
hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_ss7 0.10
hi, i made tarball with some ss7 patches from www.voip-info.org and other places and put this at http://www.freevoice.cz/chan_ss7-0.10.tgz Sifira is not in active development anymore :( (but they made good work! thanks) from Changelog New in version 0.10 (community version) - port to asterisk 1.4.14 (http://br.geocities.com/bruno_agostinho/) - added E prefix for emergency calls (www.tvtrinec.cz) - some stability fixes (www.tvtrinec.cz) - sangomazaptel example config - RBT (?) - autoPC+uptime+watermark+stats (www.ss7.pl) - cic block/unblock fix (tomasz.paszkowski at ctinf.pl) - local/remote hangup info in NOTICE (cervajs at freevoice.cz) please test and report thanks --- Marek Cervenka === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_ss7 0.10.1
hi, i'm added another patch to chan_ss7 it's from Denis Smirnov http://download.seiros.ru/SeirosPBX/chan_ss7/ New in version 0.10.1 (community version) - support for more than 256 channels - zap style addressing http://download.seiros.ru/SeirosPBX/chan_ss7/ http://www.freevoice.cz/chan_ss7/chan_ss7-0.10.1.tar.gz md5sum a3ca3031f8f4ef96d505be3b297b47cc --- Marek Cervenka === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk chan_sip tuning
hi, can i ask what settings do you recommend for a lot(1000-1) of different sip phones which are behind NAT(many different routers)? i have qualify=5000 nat=yes clisip show settings Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 asterisk 1.4 thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk optimalization
hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk optimalization
marek cervenka [EMAIL PROTECTED] writes:
hi,
i'm testing asterisk 1.4/1.2 in the following scenario
centos5/cpu quad xeon E5335 2.0Ghz
- test clients behind nat
- 1500+ testing instances - reregister option from 1min to 1hour
- qualify set to 5000
top shows over 100% cpu. cpu cores sometimes go to 95%
with htop i see ~16threads but only one child have ~95% cpu
(how i can get info about that thread? what he is doing?)
oprofile can probably tell you. It can be a bit difficult to get
all the debugging information into the right places so oprofile works,
but it's very helpful.
this is strace -p ppid_of_problematic_thread
can you look if you see any anomalies?
--cut--
socket(PF_INET, SOCK_DGRAM, IPPROTO_IP) = 14
connect(14, {sa_family=AF_INET, sin_port=htons(50195),
sin_addr=inet_addr(filtered)}, 16) = 0
getsockname(14, {sa_family=AF_INET, sin_port=htons(32777),
sin_addr=inet_addr(filtered)}, [16]) = 0
close(14) = 0
time(NULL) = 1201236754
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=806, ...}) = 0
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=806, ...}) = 0
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=806, ...}) = 0
gettimeofday({1201236754, 532554}, NULL) = 0
sendto(11, OPTIONS sip:[EMAIL PROTECTED]..., 497, 0,
{sa_family=AF_INET, sin_port=htons(5060), sin_addr=inet_addr(filtered)},
16) = 497
gettimeofday({1201236754, 532798}, NULL) = 0
gettimeofday({1201236754, 532909}, NULL) = 0
gettimeofday({1201236754, 533010}, NULL) = 0
time(NULL) = 1201236754
time(NULL) = 1201236754
gettimeofday({1201236754, 533903}, NULL) = 0
poll([{fd=11, events=POLLIN}], 1, 17) = 0
gettimeofday({1201236754, 551423}, NULL) = 0
gettimeofday({1201236754, 551535}, NULL) = 0
time(NULL) = 1201236754
time(NULL) = 1201236754
gettimeofday({1201236754, 551994}, NULL) = 0
poll([{fd=11, events=POLLIN, revents=POLLIN}], 1, 20) = 1
recvfrom(11, SIP/2.0 200 OK\r\nTo: sip:filtered..., 4095, 0,
{sa_family=AF_INET, sin_port=htons(5060), sin_addr=inet_addr(filtered)},
[16]) = 422
gettimeofday({1201236754, 557006}, NULL) = 0
gettimeofday({1201236754, 557065}, NULL) = 0
gettimeofday({1201236754, 557397}, NULL) = 0
time(NULL) = 1201236754
time(NULL) = 1201236754
time(NULL) = 1201236754
gettimeofday({1201236754, 557794}, NULL) = 0
poll([{fd=11, events=POLLIN}], 1, 14) = 0
gettimeofday({1201236754, 571604}, NULL) = 0
socket(PF_INET, SOCK_DGRAM, IPPROTO_IP) = 14
connect(14, {sa_family=AF_INET, sin_port=htons(50195),
sin_addr=inet_addr(filtered)}, 16) = 0
getsockname(14, {sa_family=AF_INET, sin_port=htons(32777),
sin_addr=inet_addr(filtered)}, [16]) = 0
close(14) = 0
time(NULL) = 1201236754
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=806, ...}) = 0
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=806, ...}) = 0
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=806, ...}) = 0
gettimeofday({1201236754, 572328}, NULL) = 0
sendto(11, OPTIONS sip:[EMAIL PROTECTED] S..., 481, 0,
{sa_family=AF_INET, sin_port=htons(5060), sin_addr=inet_addr(filtered)},
16) = 481
gettimeofday({1201236754, 572462}, NULL) = 0
gettimeofday({1201236754, 572498}, NULL) = 0
gettimeofday({1201236754, 572566}, NULL) = 0
time(NULL) = 1201236754
time(NULL) = 1201236754
gettimeofday({1201236754, 572859}, NULL) = 0
poll([{fd=11, events=POLLIN}], 1, 1)= 0
gettimeofday({1201236754, 573604}, NULL) = 0
gettimeofday({1201236754, 573651}, NULL) = 0
time(NULL) = 1201236754
time(NULL) = 1201236754
gettimeofday({1201236754, 573872}, NULL) = 0
poll([{fd=11, events=POLLIN}], 1, 5)= 0
gettimeofday({1201236754, 578602}, NULL) = 0
gettimeofday({1201236754, 578652}, NULL) = 0
time(NULL) = 1201236754
time(NULL) = 1201236754
gettimeofday({1201236754, 578863}, NULL) = 0
poll([{fd=11, events=POLLIN, revents=POLLIN}], 1, 11) = 1
recvfrom(11, \0\0\0\0, 4095, 0, {sa_family=AF_INET,
sin_port=htons(5060), sin_addr=inet_addr(filtered)}, [16]) = 4
gettimeofday({1201236754, 587424}, NULL) = 0
time(NULL) = 1201236754
time(NULL) = 1201236754
gettimeofday({1201236754, 587639}, NULL) = 0
poll([{fd=11, events=POLLIN}], 1, 2)= 0
gettimeofday({1201236754, 589599}, NULL) = 0
socket(PF_INET, SOCK_DGRAM, IPPROTO_IP) = 14
connect(14, {sa_family=AF_INET, sin_port=htons(50195),
sin_addr=inet_addr(filtered)}, 16) = 0
getsockname(14, {sa_family=AF_INET, sin_port=htons(32777),
sin_addr=inet_addr(filtered)}, [16]) = 0
close(14) = 0
time(NULL
Re: [asterisk-users] asterisk optimalization
marek cervenka [EMAIL PROTECTED] writes: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) oprofile can probably tell you. It can be a bit difficult to get all the debugging information into the right places so oprofile works, but it's very helpful. this is strace -p ppid_of_problematic_thread can you look if you see any anomalies? one more info with iptables -A INPUT -p udp -s 0/0 -d 0/0 --dport 5060 -j REJECT cpu usage goes to 0-4% - this problem is not some asterisk cpu deadlock but problem in process incoming packets --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] supermicro hardware + sangoma
hi, i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro kernels, wanpipe 3.5.6) card is: 1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36 and i have this in log irq 17: nobody cared (try booting with the irqpoll option) Pid: 0, comm: swapper Not tainted 2.6.30.9rh #1 Call Trace: [c0465b07] __report_bad_irq+0x27/0x90 [c0465caa] note_interrupt+0x13a/0x180 [c04665af] handle_fasteoi_irq+0x9f/0xd0 [c0466510] ? handle_fasteoi_irq+0x0/0xd0 IRQ [c0404506] ? do_IRQ+0x46/0xb0 [c0588234] ? acpi_hw_write_port+0x27/0x71 [c0403469] ? common_interrupt+0x29/0x30 [c05943d4] ? acpi_idle_enter_bm+0x218/0x241 [c062bf8e] ? cpuidle_idle_call+0x6e/0xc0 [c0401e45] ? cpu_idle+0x35/0x60 [c06d42f2] ? start_secondary+0x182/0x1e0 handlers: [f872d9b0] (sdla_isr+0x0/0x310 [wanpipe]) Disabling IRQ #17 dou you have idea what is the problem? irqpoll doesnt help i have tried this supermicro motherboards http://www.supermicro.com/products/motherboard/QPI/5500/X8DTU-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTi-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTL-iF.cfm do you have someone working sangoma card with Tylersburg(intel 5520/5500) chipset? thanks p.s. sorry for offtopic :( --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Asterisk 0013405]: [patch] T38 gateway (fwd)
testers needed -- Forwarded message -- Date: Wed, 11 Nov 2009 17:48:04 -0600 Subject: [Asterisk 0013405]: [patch] T38 gateway A NOTE has been added to this issue. == https://issues.asterisk.org/view.php?id=13405 == Reported By:dafe_von_cetin Assigned To: == Project:Asterisk Issue ID: 13405 Category: Applications/app_fax Reproducibility:N/A Severity: feature Priority: normal Status: confirmed Asterisk Version: SVN Regression: No Reviewboard Link: SVN Branch (only for SVN checkouts, not tarball releases): trunk SVN Revision (number only!): 140548 == Date Submitted: 2008-08-30 16:44 CDT Last Modified: 2009-11-11 17:47 CST == Summary:[patch] T38 gateway Description: Hi all, I'm sending you patch containing new application app_faxgateway.c (FaxGateway) which is able to mediate T30 to T38 and vice versa. Feature is using spands library (I used spandsp-0.0.4pre18 and spandsp-0.0.5pre4). Best regards Daniel. == -- (0113693) dafe_von_cetin (reporter) - 2009-11-11 17:47 https://issues.asterisk.org/view.php?id=13405#c113693 -- Hi, I've just uploaded the patch update for the newest trunk. The patch is still without previously mentioned transparency. Daniel. Issue History Date ModifiedUsername FieldChange == 2009-11-11 17:47 dafe_von_cetin Note Added: 0113693 == ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk cdr - remote ip address
hi, i want add info about remote party ip address to the asterisk cdr table can you recommend me the system way? thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk cdr - remote ip address
- exten = s,x,Set(CDR(userfield) = information) - replace information
with the information like ${remoteip}
${remoteip} variable doesnt exist in asterisk (for remote voip phone)
SIPURI=sip:6...@192.168.1.184:5061 doesnt have public ip
i'm only found way
- check ${CHANNEL} for name
- check astDB SIP/Registry
- set some variable
really doesnt exist some cleaner way?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk cdr - remote ip address
hi,
i want add info about remote party ip address to the asterisk cdr table
can you recommend me the system way?
---
Marek Cervenka
===
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Re: [asterisk-users] asterisk cdr - remote ip address - SOLVED
for the record
(added to http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql)
some_context
;dial trunk
exten = _X.,1,Dial(SIP/trunk/${EXTEN})
;exten h must be in same context!
exten = h,1,noop(extended CDR)
exten = h,n,set(CDR(hangupcause)=${HANGUPCAUSE}) ; hangupcause
exten = h,n,set(CDR(peerip)=${CHANNEL(peerip)}) ; like
10.0.0.5 if behind nat
exten = h,n,set(CDR(recvip)=${CHANNEL(recvip)}) ; like
194.79.52.192 - public ip
exten = h,n,set(CDR(from)=${CHANNEL(from)}) ; like
sip:1...@sip.proxy.cz
exten = h,n,set(CDR(uri)=${CHANNEL(uri)}) ; like
sip:1...@10.0.0.5
exten = h,n,set(CDR(useragent)=${CHANNEL(useragent)}) ; useragent
like Aastra_57i
exten = h,n,set(CDR(codec1)=${CHANNEL(audioreadformat)}) ; codec *
exten = h,n,set(CDR(codec2)=${CHANNEL(audiowriteformat)}) ;
exten = h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)}) ; lost
packets by local end **
exten = h,n,set(CDR(rlp)=${CHANNEL(rtpqos,audio,remote_lostpackets)}) ; lost
packets by remote end
exten = h,n,set(CDR(ljitt)=${CHANNEL(rtpqos,audio,local_jitter)}) ; the
same for jitter
exten = h,n,set(CDR(rjitt)=${CHANNEL(rtpqos,audio,remote_jitter)})
* i dont know if the same channel can have different audioreadformat and
audiowriteformat. imho not
** RTPAUDIOQOS isnt ok. check
http://lists.digium.com/pipermail/asterisk-biz/2009-November/031910.html
known problem:
it is only for caller. i dont know how to log call leg B
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk cdr - remote ip address
hi,
i want add info about remote party ip address to the asterisk cdr table
can you recommend me the system way?
---
Marek Cervenka
===
___
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[asterisk-users] ANNOUNCE: New version of Activa TAPI driver
hello, there is new version of the best open source TAPI driver for Asterisk - Activa 1.6.1 * NEW: Asterisk 1.6 compatibility (partially sponsored by IPEX a.s. http://www.ipex.cz) * NEW: FEATURE_CODES standardization for AgentACD integration login, logout, ready, notReady. * NEW: ActivaTSP x64 version. * New: Windows 2008 Server compatibility. * CHANGE: Some performance optimization. * FIX: SIP/ Dns can generate void extensions. * FIX: in process dn expresion, the duplicate filter deletes non duplicate entries. download: http://sourceforge.net/projects/activa/files/ doc: http://activa.sourceforge.net/readme.html many thanks to Activa Team --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk T.38 Gateway code testing
hi, i made page for Asterisk T.38 Gateway code testing http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming later BUT Asterisk 1.8 is too far and we need t.38 gw now if you would like help/test current code(last patch from https://issues.asterisk.org/view.php?id=13405), please contact me i have 2 public testing machines connected over E1 PLEASE do not post bug reports to https://issues.asterisk.org/view.php?id=13405 because this patch cannot be included in 1.6.2 (digium rules) i'm in contact with klaus darilion and daniel ferenci(asterisk t.38 developers) and i can arrange fixing bugs my jabber is cerv...@njs.netlab.cz look forward for better t.38 days --- Marek Cervenka jabber - cerv...@njs.netlab.cz === -- _ -- 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 T.38 Gateway code testing
asterisk t38 gw patch updated to 1.6.2.9 https://issues.asterisk.org/view.php?id=13405 i made page for Asterisk T.38 Gateway code testing http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming later BUT Asterisk 1.8 is too far and we need t.38 gw now (for testing etc) if you would like help/test current code(last patch from https://issues.asterisk.org/view.php?id=13405), please contact me i have 2 public testing machines connected over E1 my jabber is cerv...@njs.netlab.cz --- Marek Cervenka === -- _ -- 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] UDPTL T38 via NAT
On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. try asterisk 1.6.2.9 --- Marek Cervenka === -- _ -- 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] UDPTL T38 via NAT
On 06/22/2010 04:38 PM, marek cervenka wrote: On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. try asterisk 1.6.2.9 What would be the reason to do that? Is there any change on this in 1.6.2.9? yes 1.6.2.x branch is a lot better in T.38 area --- Marek Cervenka === -- _ -- 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] resending cause codes
hello,
i'm testing sending ISDN cause codes to customer pbx (test scenario for
unallocated number)
topology:
PSTN-E1-AsteriskA-AsteriskB-SOMEPBX
INVITE from SOMEPBX to PSTN
AsteriskA sends to AsteriskB
Status-Line: SIP/2.0 503 Service Unavailable
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
how can i resend HangupCauseCode from AsteriskB to SOMEPBX?
i'm tried this on AsteriskB
exten = _X.,1,Dial(SIP/AsteriskA,${EXTEN})
exten = _X.,n,Hangup(${SIP_HEADER(X-Asterisk-HangupCauseCode)})
thanks
--
---
Marek Cervenka
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Re: [asterisk-users] Sharing Fail2ban data
I've been doing a little work that I wanted to share. We've had a number of Asterisk systems that have been under heavier than normal attack. We use fail2ban but we either have to let each system be exposed or keep all the data synchronized which is a bit of a chore. I wrote a little server that assists in keeping data synchronized across sites. If you're interested in using it to assist in managing your own fail2ban sharing list I'll gladly share it. I also am offering it as a free service for those who are interested in contributing to a blacklist. If you're interested the information is available here: http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in the server code just drop me an email. i'm interested in the server code. thanks -- --- Marek Cervenka === -- _ -- 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] sip trunk balancing
hi, is there some way to balance accross sip trunks by the number of calls? example 3 trunks alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3) alfa have 25 calls now i want next call terminate to delta. how to find in asterisk the current calls number on sip trunk alfa? 1) set call-limit in sip.conf. then in the dialplan sip show peer inuse|grep alfa - parse - if numcalls 25 then dial(sip/delta) 2) groupcount ? 3) what else? thanks Marek -- _ -- 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 rpm build problem
hi, i'm trying build asterisk rpm normal compilation is ok but rpm building always fail centos6/asterisk 1.8.5.0 any ideas? gcc -o recno/rec_utils.o -c recno/rec_utils.c -MD -MT recno/rec_utils.o -MF .recno_rec_utils.o.d -MP -pthread -Wall -D__DBINTERFACE_PRIVATE -I. -I.. -Iinclude -Ihash -Ibtree -Irecno -I/root/rpmbuild/BUILD/asterisk-1.8.5.0/include -O2 -g -march=i386 -mtune=i686 -Werror-implicit-function-declaration -I/usr/include/libxml2 -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations-Wno-strict-aliasing -O2 -g -march=i386 -mtune=i686 -Werror-implicit-function-declaration ar cr libdb1.a hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o ranlib libdb1.a make[2]: Leaving directory `/root/rpmbuild/BUILD/asterisk-1.8.5.0/main/db1-ast' /root/rpmbuild/BUILD/asterisk-1.8.5.0/build_tools/make_linker_version_script asterisk gcc -o asterisk -Wl,--export-dynamic -Wl,--version-script,asterisk.exports -Wl,--dynamic-list,asterisk.dynamics abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o db1-ast/libdb1.a buildinfo.o -lssl -lcrypto -lc -lxml2 -lz -lm -ldl -lpthread -ltermcap -lm -lresolv -ledit -lcurses astobj2.o: In function `ast_atomic_fetchadd_int': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' ccss.o: In function `ast_atomic_fetchadd_int': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' cdr.o:/root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:646: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make[1]: Leaving directory `/root/rpmbuild/BUILD/asterisk-1.8.5.0/main' make: *** [main] Error 2 error: Bad exit status from /var/tmp/rpm-tmp.FeglRm (%build) -- --- Marek Cervenka === -- _ -- 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] call pickup
hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --- Marek Cervenka === -- _ -- 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] call pickup
On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some NOTIFY to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without monitoring the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- 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] cdr documentation - new fields
hi, there are 3 new cdr fields in asterisk 1.8 (https://wiki.asterisk.org/wiki/display/AST/New+in+1.8#Newin1.8-CDR) linkedid - is based on uniqueID, but spreads to other channels as transfers, dials, etc are performed. Thus the pieces of CDR can be grouped into multilegged sets. sequence - can be combined with linkedid or uniqueid to uniquely identify a CDR. peeraccount - ? can someone with write permissions fix this doc? https://wiki.asterisk.org/wiki/display/AST/CDR+Fields https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend thanks -- --- Marek Cervenka === -- _ -- 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] Calendar Integration Problem
Dne 30.4.2012 11:09, Bharat Lalcheta napsal(a): Hiii all, I am using asterisk 1.8.9.2 and compile all modules related to calendar. neon version is 0.29.6. OS is ubuntu 11.10. I configured ical for zimbra, caldav for google mail and ews for exchange 2010 calendar. ical and caldav setup working fine and i am getting my calendar events perfectly. But for exchange 2010 calendar i am getting following error. Unable to communicate with Exchange Web Service at 'https://ex1.domain.com/EWS/Exchange.asmx': Could not authenticate to server: ignored NTLM challenge, GSSAPI authentication error: Unspecified GSS failure. Minor code may provide more information: Credentials cache file '/tmp/krb5cc_0' not found my calendar.conf is as follows [calendar3] type = ews ; type of calendar--currently supported: ical, caldav, exchange, or ews url = https://ex1.domain.com/EWS/Exchange.asmx ; URL to MS Exchange EWS user = myn...@domain.com mailto:myn...@domain.com ; Exchange username secret = xx ; Exchange password refresh = 10 ; refresh calendar every n minutes timeframe = 20 try user = domain.com/myname mailto:myn...@domain.com ; Exchange username -- _ -- 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] axfer with simple CDR
hi, i read a lot about CDR problems this document is the best description of CDRs problem in Asterisk http://svn.digium.com/svn/asterisk/team/murf/RFCs/CDRfix2.rfc.docx i found but i cant still answer my question is it possible with simple CDR fully describe axfer? (axfer is asterisk native, not phone function) scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) (what about ring time?) is it possible? if yes, can you post some example? -- --- Marek Cervenka === -- _ -- 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] axfer with simple CDR
Dne 29.5.2012 18:23, Kevin P. Fleming napsal(a): On 05/29/2012 07:57 AM, Marek Cervenka wrote: is it possible with simple CDR fully describe axfer? (axfer is asterisk native, not phone function) No, it is not. CDRs (Asterisk or otherwise) are only capable of directly (simply) describing a call from party A to party B. They have no ability to describe call treatments, in-call features, or any other advanced features. Asterisk's CDRs *attempt* to represent such information, but as you've seen, they don't satisfy everyone, and it seems that many parties have conflicting ideas as to how things like transfers should be represented in CDRs. ok ok. i tried it :) i'll try it the right way - CEL (centos6,unixODBC,cel_odbc,mysql) any sql views,scripts,sql triggers someone? is it implemented in switchvox,asterisknow,trixbox,elastix? -- --- Marek Cervenka === -- _ -- 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] attended transfer with CEL
hello, is there someone who successfully get info about attended transfer from CEL? if yes, can you post some hints/algorithm/...? scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka === -- _ -- 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] attended transfer with CEL
https://wiki.asterisk.org/wiki/display/AST/CEL+Function on this wiki is THIS IS NO LONGER TRUE REWRITE is there some way to write userfield,accountcode to the cel? Dne 5.6.2012 13:21, Marek Cervenka napsal(a): hello, is there someone who successfully get info about attended transfer from CEL? if yes, can you post some hints/algorithm/...? scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka === -- _ -- 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] attended transfer with CEL
Dne 20.6.2012 18:40, Marek Cervenka napsal(a): https://wiki.asterisk.org/wiki/display/AST/CEL+Function on this wiki is THIS IS NO LONGER TRUE REWRITE is there some way to write userfield,accountcode to the cel? solved. it's set(CHANNEL(userfield)=something) another question i'm using patch from https://issues.asterisk.org/jira/browse/ASTERISK-18037 it works great but there is problem(bug?) in second axfer A - call - B - axfer(AtoC) - C - axfer(AtoD) D in cel is eventtype, cid_num, exten HOLD_START, A, B HOLD_STOP, A, B BUT second axfer is HOLD_START, B, C HOLD_STOP, B, C this is strange because on hold is A. is it a bug? very big problem is that, i cant get info about A - D call (after second axfer). there is no info about bridged channel A after axfer Dne 5.6.2012 13:21, Marek Cervenka napsal(a): hello, is there someone who successfully get info about attended transfer from CEL? if yes, can you post some hints/algorithm/...? scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerve...@slu.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE,RHCVA 100-175-678 === -- _ -- 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] Proactive problem monitoring on SIP on Asterisk
Dne 21.6.2012 9:52, Ishfaq Malik napsal(a): On Wed, 2012-06-20 at 20:04 +0200, Stefan at WPF wrote: Hello, 1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip debug) log and tells me on which of them were problems? Checking the logs manually is very hard, but as SIP is a standardized protocoll, there should be tools doing that for you? As an example, a person calling me recently got a 488 Not acceptable error as reply from my Asterisk box. Nothing came through to my SIP phone, so I didn't know anything about the call or the problems (which were on his phone btw). I would like to be informed about such cases, know that there was a call to my Asterisk box that made problems. 2) How about monitoring speech quality? E.g. sometimes it seems like a packet is missing (I then have a short pause during the call), how to monitor such things and create statistics out of this data? So basically I want to monitor my Asterisk installation proactively for reliability/problems and (speech) quality. check asterisk testsuite https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation thereis scenarios for console sip client pjsua(from pjproject) which can perform speech quality measurement marek cervenka -- _ -- 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] AGI not generating sip 180/183 status
hello, i have strange problem with AGI (asterisk 1.8.10.0) when i use Dial from dialplan everything is ok when i dial from AGI script there is missing SIP Status 180 ringing and 183 session progress any ideas? DIAL without AGI 196.356479 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 196.356768 10.0.0.213 - 10.0.0.193 SIP Status: 401 Unauthorized 196.365709 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333...@some.pbx.org 196.370028 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 196.370503 10.0.0.213 - 10.0.0.193 SIP Status: 100 Trying 199.797325 10.0.0.213 - 10.0.0.193 SIP Status: 180 Ringing 199.797932 10.0.0.213 - 10.0.0.193 SIP/SDP Status: 183 Session Progress, with session description 199.878441 10.0.0.193 - 10.0.0.213 RTCP Receiver Report Source description 199.988259 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xD2C6DEB8, Seq=7289, Time=3171500, Mark 200.004139 10.0.0.213 - 10.0.0.193 RTP PT=ITU-T G.711 PCMA, SSRC=0x279E385A, Seq=50775, Time=28960 200.008118 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xD2C6DEB8, Seq=7290, Time=3171660 201.504218 10.0.0.213 - 10.0.0.193 RTP PT=ITU-T G.711 PCMA, SSRC=0x279E385A, Seq=50850, Time=40960 201.519477 10.0.0.193 - 10.0.0.213 SIP Request: BYE sip:222333444@10.0.0.213:5060 201.519611 10.0.0.213 - 10.0.0.193 SIP Status: 487 Request Terminated 201.519800 10.0.0.213 - 10.0.0.193 SIP Status: 200 OK 201.528465 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333...@some.pbx.org DIAL from AGI 66.581752 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 66.581958 10.0.0.213 - 10.0.0.193 SIP Status: 401 Unauthorized 66.590738 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333...@some.pbx.org 66.59 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 66.596167 10.0.0.213 - 10.0.0.193 SIP Status: 100 Trying 66.652571 10.0.0.213 - 10.0.0.193 SIP/SDP Status: 200 OK, with session description 66.676485 10.0.0.193 - 10.0.0.213 RTCP Receiver Report Source description 66.750371 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333444@10.0.0.213:5060 66.844392 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq=3869, Time=1120100, Mark 66.854430 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq=3870, Time=1120260 ... 69.404625 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq=3998, Time=1140740 69.516390 10.0.0.193 - 10.0.0.213 SIP Request: BYE sip:222333444@10.0.0.213:5060 69.516669 10.0.0.213 - 10.0.0.193 SIP Status: 200 OK -- --- Marek Cervenka === -- _ -- 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] salesforce opencti
hello, do you have someone connector to salesforce? http://wiki.developerforce.com/page/Open_CTI i need it for Mac OS X (the web/javascript way, not the old CTI Adapter way) i'm using Asterisk 1.8 thanks -- --- Marek Cervenka === -- _ -- 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] WebM / VP8 support
hello, any news about WebM/VP8 support in asterisk? some bounty where can i contribute? -- --- Marek Cervenka === -- _ -- 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] sip video endpoint with asterisk
hi, i need some small sip video endpoint for cloud videoconference (like bluejeans) i have this idea VIDEO OUT TV or projector with HDMI VIDEO IN cameras with h264 hw enconding - http://downloads.element14.com/raspberry-pi-camera/ http://downloads.element14.com/raspberry-pi-camera/ - logitech C920 - Creative Live! Cam Connect HD - ??? ENDPOINT - raspberry - miniPC linux + asterisk ? https://wiki.asterisk.org/wiki/display/AST/Video+Console https://wiki.asterisk.org/wiki/display/AST/Video+Console AUDIO IN + AUDIO OUT microphone with integrated speakers for the table http://www.jabra.com/Products/PC_Headsets/Jabra_SPEAK__510_Series/Jabra_Speak_510 http://www.jabra.com/Products/PC_Headsets/Jabra_SPEAK__510_Series/Jabra_Speak_510 (bluetooth connection!!!) http://www.phnxaudio.com/quattro3 http://www.phnxaudio.com/quattro3 http://www.yamaha.com/products/en/communication/usb_conference_speakerphones/ http://www.yamaha.com/products/en/communication/usb_conference_speakerphones/ http://www.dev-audio.com/products/microcone/ http://www.dev-audio.com/products/microcone/ http://www.clearone.com/products_chat160 http://www.clearone.com/products_chat160 do you think it is possible? any recommendations? -- --- Marek Cervenka === -- _ -- 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] groupcount fraud problem
hi, i have strange problem with call-limit/groupcount limiting. i set up limit of 2 calls. i'm using both methods but a for few times i have problem with successfull fraud with more calls than 2 asterisk is 1.8.22 someone with the same problem? any ideas how to solve or debug this problem? -- --- Marek === -- _ -- 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] groupcount fraud problem
Dne 14.8.2013 13:35, Marek Cervenka napsal(a): hi, i have strange problem with call-limit/groupcount limiting. i set up limit of 2 calls. i'm using both methods but a for few times i have problem with successfull fraud with more calls than 2 asterisk is 1.8.22 someone with the same problem? any ideas how to solve or debug this problem? it's seems like they are using some transfer or system code to modify call flow in CDR i see | calldate | duration| billsec | peerip | recvip | useragent | uniqueid | uri | 2013-08-10 17:12:52 |7 | 2 | attacker_ip | attacker_ip | eyeBeam release 3006o stamp 17 | 1375679572.17728 | sip:clid_number@attacker_ip:14932 | | 2013-08-10 17:13:03 | 666 | 660 | siptrunk_ip | siptrunk_ip | operator_switch | 1375679583.17730 | sip:called_number@siptrunk_ip:5060 | -- --- Marek === -- _ -- 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] mixmonitor extension
hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --- Marek Cervenka === -- _ -- 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] mixmonitor extension
can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know Dne 23.1.2014 16:31, Gareth Blades napsal(a): On 23/01/14 15:21, Marek Cervenka wrote: hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? core show file formats will give you a list of formats your system supports together with the filename extension. Not all may be supported for writing (mp3 being one example I believe). core show file formats Format Name Extensions -- -- slin mp3mp3 h264 h264 h264 g729 g729 g729 g719 g719 g719 gsmgsmgsm g726 g726-16g726-16 g726 g726-24g726-24 g726 g726-32g726-32 g726 g726-40g726-40 h263 h263 h263 gsmwav49 WAV|wav49 g722 g722 g722 ulaw au au alaw alaw alaw|al|alw ulaw pcmpcm|ulaw|ul|mu|ulw siren14siren14siren14 siren7 siren7 siren7 slin192sln192 sln192 slin96 sln96 sln96 slin48 sln48 sln48 slin44 sln44 sln44 slin32 sln32 sln32 slin24 sln24 sln24 slin16 sln16 sln16 slin12 sln12 sln12 slin slnsln|raw slin16 wav16 wav16 slin wavwav g723 g723sf g723|g723sf ilbc iLBC ilbc 30 file formats registered. -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerve...@slu.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE,RHCVA 100-175-678 === -- _ -- 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] mixmonitor extension
i'm talking about native mp3,opus support in mixmonitor application. read the first answer from Gareth Blades Dne 24.1.2014 1:39, Patrick Lists napsal(a): On 24-01-14 00:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later versions of asterisk you can enable format_mp3 in make menuselect. what about patch for Opus? uncle google doesnt know Did you really google? http://lmgtfy.com/?q=asterisk+opus Regards, Patrick -- --- Marek Cervenka === -- _ -- 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] mixmonitor extension
for the record. info about opus from Lorenzo Mniero (author of Opus patch for asterisk) with his permission --cite-- Opus is just a codec. In order to save an audio file using Opus, you need a container, which for Opus is OGG. Asterisk supports OGG, but I think it is implemented to only dump Vorbis audio, and so the existing module would need to be extended to support Opus as well. I haven't checked how complex this could be, to be honest, so I have no idea about how much effort would be needed for this. Right now we don't need it, so I really can't say if and when we'll start working on this. Lorenzo --cite-- Dne 24.1.2014 10:42, Gareth Blades napsal(a): On 23/01/14 23:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know MP3 is only supported for reading not writing. Its a patent issue as Asterisk cannot distribute the software to write to mp3 under its own license. Its a similar issue with Opus as the codec is covered by a couple of patents in the USA. What most people do is use MixMonitor to record to .wav (alaw) and then in the 'h' extension call a program which runs a background task to convert the .wav file to whatever format they wish. Thats what we do but we actually use the Monitor application and we end up with both legs of the call and multiple sets of recordings if people pause and unpause. We then move these files off to a different server when they get mixed and converted to mp3 and then emailed out to our customers. We do it this way to reduce the load on the Asterisk boxes but also keep all the call recordings in a central location. -- --- Marek Cervenka === -- _ -- 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 SugarCrm integration
hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- --- Marek Cervenka === -- _ -- 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 SugarCrm integration
it's old. sugarcrm v7 is not supported Dne 28.8.2014 v 14:54 Scott Griepentrog napsal(a): I've used this before, and it appears to still be an active project. https://github.com/blak3r/yaai On Thu, Aug 28, 2014 at 7:29 AM, Marek Cervenka cerv...@fpf.slu.cz mailto:cerv...@fpf.slu.cz wrote: hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- _ -- 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 multiple ip
hi, i need migrate customers from severeal to one asterisk server with multiple ip aliases like eth0 192.168.10.1 eth0:1 192.168.10.20 eth0:2 192.168.10.30 i must preserve endpoint configuration to these ip adressess the problem is if i register to 192.168.10.30, the answer is from 192.168.10.1 are there some ways for this scenario? 1) chan_pjsip? 2) kamailio in front of asterisk on the same server? 3) iptables magic? 4) ... thanks -- --- Marek Cervenka === -- _ -- 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 multiple ip
it looks like i found solution with chan_pjsip /etc/asterisk/pjsip.conf [transport-udp-net1] type=transport protocol=udp bind=192.168.10.20 [transport-udp-net2] type=transport protocol=udp bind=192.168.10.30 [net1_user1] type=endpoint transport=transport-udp-net1 [net2_user1] type=endpoint transport=transport-udp-net2 can you someone confirm this solution? Dne 29.8.2014 v 11:26 Marek Cervenka napsal(a): hi, i need migrate customers from severeal to one asterisk server with multiple ip aliases like eth0 192.168.10.1 eth0:1 192.168.10.20 eth0:2 192.168.10.30 i must preserve endpoint configuration to these ip adressess the problem is if i register to 192.168.10.30, the answer is from 192.168.10.1 are there some ways for this scenario? 1) chan_pjsip? 2) kamailio in front of asterisk on the same server? 3) iptables magic? 4) ... thanks -- --- Marek Cervenka === -- _ -- 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] opus 11.12.0
hi, any plans update patch for 11.12.0? |https://github.com/meetecho/asterisk-opus https://github.com/netaskd/asterisk-opus/ | patching file build_tools/menuselect-deps.in patching file channels/chan_sip.c Hunk #1 succeeded at 7659 (offset -98 lines). Hunk #2 succeeded at 11011 (offset -34 lines). Hunk #3 succeeded at 11050 (offset -34 lines). Hunk #4 succeeded at 7 with fuzz 1 (offset -34 lines). Hunk #5 FAILED at 12722. 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file codecs/codec_opus.c patching file codecs/ex_opus.h patching file configure.ac Hunk #2 succeeded at 2150 (offset 31 lines). patching file formats/format_vp8.c patching file include/asterisk/format.h patching file main/channel.c patching file main/format.c Hunk #6 succeeded at 1098 (offset 12 lines). patching file main/frame.c patching file main/rtp_engine.c Hunk #1 succeeded at 2326 (offset 37 lines). Hunk #2 succeeded at 2370 (offset 37 lines). patching file makeopts.in patching file res/res_rtp_asterisk.c Hunk #1 succeeded at 95 with fuzz 1 (offset 4 lines). Hunk #2 FAILED at 349. Hunk #3 succeeded at 3011 (offset 394 lines). Hunk #4 succeeded at 3097 (offset 394 lines). 1 out of 4 hunks FAILED -- saving rejects to file res/res_rtp_asterisk.c.rej thanks -- --- Marek Cervenka === -- _ -- 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] Tutorial: compiling and installing Asterisk 13
Dne 12.9.2014 v 11:27 Lenz Emilitri napsal(a): Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. you can shrink it by: - srtp is in EPEL repo [root@dev6 ~]# yum list|grep srtp libsrtp.i686 1.4.4-4.20101004cvs.el6@epel libsrtp-devel.i686 1.4.4-4.20101004cvs.el6@epel - jansson is in EPEL repo [root@dev6 ~]# yum list|grep jansson jansson.i686 2.6-1.el6 @epel jansson-devel.i686 2.6-1.el6 @epel - pjproject spec file https://bugzilla.redhat.com/show_bug.cgi?id=1140324 -- --- Marek Cervenka === -- _ -- 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
