[asterisk-users] DAHDI 2.7.0.1 and CentOS 6.5
Hello,
during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error:
In file included from
/usr/src/dahdi-linux-2.7.0.1/drivers/dahdi/dahdi-base.c:66:
/usr/src/dahdi-linux-2.7.0.1/include/dahdi/kernel.h:1407: error:
redefinición de 'PDE_DATA'
include/linux/proc_fs.h:328: nota: la definición previa de 'PDE_DATA'
estaba aquí
make[2]: *** [/usr/src/dahdi-linux-2.7.0.1/drivers/dahdi/dahdi-base.o]
Error 1
make[1]: *** [_module_/usr/src/dahdi-linux-2.7.0.1/drivers/dahdi] Error 2
make[1]: se sale del directorio `/usr/src/kernels/2.6.32-431.el6.x86_64'
make: *** [modules] Error 2
I don't know if is the right wayt to solve it but in the:
nano include/dahdi/kernel.h
I commented out these lines:
/*static inline void *PDE_DATA(const struct inode *inode)
{
return PDE(inode)-data;
}
*/
then make and make install work.
I think the problem is there is similar declaration on the linux-kernel
source, file:
/usr/src/kernels/2.6.32-431.el6.x86_64/include/linux/proc_fs.h
Regards
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Re: [asterisk-users] DAHDI 2.7.0.1 and CentOS 6.5
Hello, thank you for the information. I'll wait the new release. Regards El 02/12/2013 16:21, Patrick Lists escribió: On 12/02/2013 10:09 PM, Bakko wrote: Hello, during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error: [snip] This was discussed earlier today and Russ pointed to the fixes: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=5ec9d756aac1a0eb5c1f48eb110e80946b43f41a https://issues.asterisk.org/jira/browse/DAHLIN-330 The fix will be in 2.8.0-rc3. Either wait for the rc3 or add the patch to your build (don't know if it works on 2.7.0.1). Regards, Patrick -- _ -- 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 11.6.0 not starting up
Hi Elder, on Linode VPS, when you execute make menuselect, on Compiler Flags menu you have to deselect BUILD NATIVE parameter. Then make, make install, make samples, make config Regards El 25/11/2013 11:49, Leandro Dardini escribió: On which kind of processor are you trying to run asterisk? Is it a real or emulated CPU? Leandro 2013/11/25 Daniel - Asterisk earohua...@gmail.com mailto:earohua...@gmail.com Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with asterisk -vvc and service asterisk start. Starting process just stop and shows: Illegal instruction as final output. Looking at logs I fouind at /var/log/asterisk/messages : [Nov 25 11:09:26] Asterisk 11.6.0 built by root @ (my-pbx-server) on a i686 running Linux on 2013-11-25 15:10:00 UTC [Nov 25 11:09:26] NOTICE[24118] cdr.c: CDR simple logging enabled. [Nov 25 11:09:26] NOTICE[24118] loader.c: 205 modules will be loaded. [Nov 25 11:09:26] NOTICE[24118] res_odbc.c: res_odbc loaded. [Nov 25 11:09:26] NOTICE[24118] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine curl [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine odbc [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine sqlite3 Any help would be welcome. My Linux distro is: Linux (my-ip-address) 3.11.6-x86-linode54 #1 SMP Wed Oct 23 15:22:49 EDT 2013 i686 GNU/Linux Elder D. Arohuanca Lima - Peru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.6.0 not starting up
Hello Elder, I don't remember where but I read in some place on virtualized servers you have to deselect this parameter. I'm using Linode too. Regards El 25/11/2013 13:16, Daniel - Asterisk escribió: Hello Bakko: Asterisk 11 is working now. Would that selection on COMPILER FLAGS needed for al IaaS plattforms? Thank you! Elder D. Arohuanca Lima - Peru On Mon, Nov 25, 2013 at 12:07 PM, Bakko asannu...@gmail.com mailto:asannu...@gmail.com wrote: Hi Elder, on Linode VPS, when you execute make menuselect, on Compiler Flags menu you have to deselect BUILD NATIVE parameter. Then make, make install, make samples, make config Regards El 25/11/2013 11:49, Leandro Dardini escribió: On which kind of processor are you trying to run asterisk? Is it a real or emulated CPU? Leandro 2013/11/25 Daniel - Asterisk earohua...@gmail.com mailto:earohua...@gmail.com Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with asterisk -vvc and service asterisk start. Starting process just stop and shows: Illegal instruction as final output. Looking at logs I fouind at /var/log/asterisk/messages : [Nov 25 11:09:26] Asterisk 11.6.0 built by root @ (my-pbx-server) on a i686 running Linux on 2013-11-25 15:10:00 UTC [Nov 25 11:09:26] NOTICE[24118] cdr.c: CDR simple logging enabled. [Nov 25 11:09:26] NOTICE[24118] loader.c: 205 modules will be loaded. [Nov 25 11:09:26] NOTICE[24118] res_odbc.c: res_odbc loaded. [Nov 25 11:09:26] NOTICE[24118] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine curl [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine odbc [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine sqlite3 Any help would be welcome. My Linux distro is: Linux (my-ip-address) 3.11.6-x86-linode54 #1 SMP Wed Oct 23 15:22:49 EDT 2013 i686 GNU/Linux Elder D. Arohuanca Lima - Peru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-12 issue after successful installation
Hello, is a 64bit installation? Maybe Asterisk looking for the file on /usr/lib64 and you have this file on /usr/lib. In this case create a symbolic link to /usr/lib64 Regards El 21/10/2013 07:26, virendra bhati escribió: Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory [root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com mailto:virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) View my profile on LinkedIn http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bind to ipv4 ipv6
This means i can't use IPv4 and IPv6 together. Right? El 27/09/2013 11:25, Eric Wieling escribió: From sip.conf.sample included in your Asterisk source tree. See item c) and the Note: ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for ; udpbindaddr, tcpbindaddr, and tlsbindaddr.) ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) ; ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 ; for TLS). ; IPv4 example: bindaddr=0.0.0.0:5062 ; IPv6 example: bindaddr=[::]:5062 ; ; The address family of the bound UDP address is used to determine how Asterisk performs ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only ; records are considered. In case d), both A and records are considered. Note, ; however, that Asterisk ignores all records except the first one. In case d), when both A ; and records are available, either an A or record will be first, and which one ; depends on the operating system. On systems using glibc, records are given ; priority. udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel van den Berg Sent: Friday, September 27, 2013 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to bind to ipv4 ipv6 Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 05:59 PM, Johan Wilfer wrote: http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.htm l Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for Asterisk+Pacemaker+Corosync+DRBD example
Hello Edwards you can install fedora repositories and the HeartBeat from those repositories. If the failover is only for two servers, this is a good solution. In the directory list, you have to add /etc/dahdi (is you use dahdi) and /var/spool/asterisk Regards El 19/09/2013 08:58, Steve Edwards escribió: I'm trying to setup a pair of FreePBX-4.211.64 boxes using Pacemaker, Corosync, and DRBD. All the examples I've found so far use Heartbeat, but Heartbeat is not in the repositories and doesn't want to compile from source. Does anyone have a working configuration they can share or a tutorial they can point me to? Also, what does drbdlinks bring to the party? Isn't just linking the 'top level' directories (/etc/asterisk/, /var/lib/asterisk/, /var/lib/mysql, etc) sufficient? -- _ -- 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 12 issue
Hello, my fault. I used pjsip oficial versión. With the git version all it's ok. Asterisk 12 start fine with pjsip stack. Now some tests. Thank you regards El 03/09/2013 10:10, Joshua Colp escribió: Bakko wrote: hello, Greetings, I' trying to use Asterisk 12 Alpha. Compilation and instalation without issues. When I try to start asterisk with: asterisk -cvvv i see this error on the console: 17:09:43.559 sip_endpoint.c !Module mod-refer registered asterisk: ../src/pjsip-simple/evsub.c:415: pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. Any hints? There's three possible reasons for this: 1. The new SIP modules have been linked statically which means each module has an independently operating copy. As things are written to operate as a whole this can fail miserably. To see if this is the case you can run: ldd /usr/lib/asterisk/modules/res_pjsip.so If the output contains no reference to libpj.so then that is your problem and you will need to follow the instructions on the wiki to remove an old pjproject. 2. The res_sip_pubsub.so module is not being loaded or has not been built. You can check the console output when loading to see if this is the case, although the ultimate reason may be below this. 3. A slight derivative on the above is that the module will be loaded, but something is trying to use it before hand. You can manually modify your modules.conf to have an explicit load order for the new SIP modules. This can require some trial/error. If this resolves the issue then we need to adjust things to make it happen automagically. As such if this is the case please open a JIRA issue so we can ensure others do not run into the same issue. Cheers, PS: Thanks for giving Asterisk 12 a go and sorry you ran into this problem! -- _ -- 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 12 issue
hello, I' trying to use Asterisk 12 Alpha. Compilation and instalation without issues. When I try to start asterisk with: asterisk -cvvv i see this error on the console: 17:09:43.559 sip_endpoint.c !Module mod-refer registered asterisk: ../src/pjsip-simple/evsub.c:415: pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. Any hints? Thank you Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk 11 on VirtualBox: Illegal Instruction
Hello, enter in make menuselect - Compiler flags and disable BUILD_NATIVE option; then recompile Asterisk Regards El 06/06/2013 10:12, jorgeart...@protoboardmx.com escribió: I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two guides. The first is the one from Asterisk: The Definitive Guide 4th edition (http://ofps.oreilly.com/titles/9781449332426/asterisk-Install.html) and the one from Billy Chia How to Install Asterisk 11 on Ubuntu 12.04 LTS (http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/). I'm able to install Dahdi, Libpri and Asterisk with no errors but as soon as I try to start asterisk with: /etc/init.d/asterisk start I got an error: Illegal Instruction (coredump). For what I have read this might be because Asterisk isn't compiling for the right architecture but I don't know how to solve this issue. Hope you can give me some guidance here. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net http://myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn after in extconfig.conf: sipusers = mysql,general,Comptes_SIP sippeers = mysql,general,Comptes_SIP iaxusers = mysql,general,Comptes_IAX iaxpeers = mysql,general,Comptes_IAX extensions = mysql,general,Extensions meetme = mysql,general,MeetMe musiconhold = mysql,general,Musiconhold voicemail = mysql,general,VoiceMail and in cdr_mysql.conf [global] hostname=myhost.mydomain.net http://myhost.mydomain.net dbname=MyDB table=Cdr password=MyPassword user=MyUser port=3306 sock=/tmp/mysql.sock [aliases] start=calldate end=callend callerid=clid src=src dst=dst dcontext=dcontext channel=channel dstchannel=dstchannel lastapp=lastapp lastdata=lastdata duration=duration billsec=billsec disposition=disposition amaflags=amaflags accountcode=accountcode userfield=userfield uniqueid=uniqueid CodeTier=CodeTier you know what file I forgot to configure? Olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com mailto:o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com http://myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com http://myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table
[asterisk-users] Fwd: Google Calendar issue
From 23 juanary 2013 on calendar.conf have to change type=caldav to type=ical http://forums.asterisk.org/viewtopic.php?f=1t=85623 http:// Regards Mensaje original Asunto: Google Calendar issue Fecha: Sat, 23 Feb 2013 10:22:17 -0500 De: Bakko asannu...@gmail.com Para: asterisk-users@lists.digium.com hello, I'm trying to connect Asterisk to Google Calendar. The connection work fine but Asterisk don't retrieve any programmed event present on the calendar. Asterisk version 1.8.20.1 Any hint? Thank you - Bakko -- _ -- 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] Google Calendar issue
hello, I'm trying to connect Asterisk to Google Calendar. The connection work fine but Asterisk don't retrieve any programmed event present on the calendar. Asterisk version 1.8.20.1 Any hint? Thank you - Bakko -- _ -- 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 not return int value
Hello,
you can use SNMP Asterisk integration to monitoring the actives calls
(Total calls, SIP calls, IAX2 calls, DAHDI calls)
Regards
El 16/02/2013 08:04, Farooq Hussain escribió:
Hello Everyone,
I have write a script following script for nagios
-- typeset -i CRITICAL;
#Positional parameter
CRITICAL=`echo $2`;
ME=`basename $0`;
#echo $CRITICAL
if [[ $2 == ]]
then
echo NO INPUT!!! Usage ./$ME -c N
else
typeset -i ASCALLS;
ASCALLS=`asterisk -rx core show channels | grep active | grep call |
awk '{print $1}'`
#echo $ASCALLS;
#mload=`echo $ASCALLS | $BC`;
#echo $((num+1))
if [[ $ASCALLS -lt $CRITICAL ]]
then
echo OK!!! Total Active Calls:$ASCALLS;
exit 0
else
echo CRITICAL!!! Total Active Calls:$ASCALLS;
exit 2
fi
fi
Thanks
But following in not return int value
`asterisk -rx core show channels | grep active | grep call | awk
'{print $1}'`
Please let me know if anyone help me in regard
Farooq Hussain
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[asterisk-users] SayDigits
Hello Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Reagards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SayDigits
Hello,
My final solution:
...
same = n,Gosub(dati,s,1(${card}))
[dati]
exten = s,1,NoOp
same = n,Set(say=${LEN(${ARG1})})
same = n,Set(digit=0)
same = n,While($[${digit} ${say}])
same = n,Saydigits(${ARG1:${digit}:1})
same = n,Wait(.75)
same = n,Set(digit=$[${digit} + 1])
same = n,Endwhile
same = n,Return
Thank you for yours suggestion
regards
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Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
me too. regards El 16/01/2013 13:25, Eric Wieling escribió: I am also experiencing this issue. Asterisk is in fact running, you can verify by running asterisk -rvvv (-r connects to an EXISTING asterisk process) or using ps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Wednesday, January 16, 2013 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the afternoon, I got the notification that asterisk 1.8.20.0 had been released, so today, I downloaded the latest 1.8-current.tar.gz and compiled and installed it (./configure, make menuselect and choose all the same options as my previous install, make, make install). Now, when I start the asterisk service using service asterisk start from the command line, this is the output: [root@pbx ~]# service asterisk start Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Starting asterisk: However, the /var/run/asterisk/asterisk.ctl file is being created and the process is starting: [root@pbx ~]# ls -lh /var/run/asterisk/ total 4.0K srwxr-xr-x 1 root root 0 Jan 16 12:07 asterisk.ctl -rw-r--r-- 1 root root 6 Jan 16 12:07 asterisk.pid However, I'm no longer getting the usual splash message when I connect to the asterisk console...this is what I get: [root@pbx ~]# asterisk -r Verbosity is at least 3 pbx*CLI I don't have any peers setup yet, or even any dialplan configured to test, but when I go through the logs, I don't find any errors or warnings that I'm not expecting. I've gone back to the asterisk 1.8.19.1 install and everything works as expected (no error messages, full splash about license / version on connection to console, etc). I performed make clean in my 1.8.20 source directory, then ./configure, make menuselect, make, make install, and even make config, and I'm still seeing this message pop up when restarting / starting the service. I went through the CHANGELOG.TXT for 1.8.20.0 and it appears there are some items talking about changing the way the process starts up (commit r376428), but I'm not enough of a coder to understand if those would cause what I'm seeing. Is anyone else seeing this issue? Should I open an issue on the tracker? Anyone see something obvious I missed? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 parking in a multi-tenant system
Hello, from 1.6.2 version, Asterisk suport multi-tenant parking Look at features.conf for a example. Regards El 15/01/2013 15:58, Carlos Alvarez escribió: We use Asterisk as a hosted PBX. We've had a couple of requests for parking, but none of the documentation shows any way to make it aware of contexts or otherwise make it multi-tenant. Have I missed something and does anyone know how to make this work? Would be on Asterisk 1.6 for now, 1.8 some time soon. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installation Problem with asterisk 1.6
yum install libxml2-devel El 03/11/2012 13:55, akhilesh chand escribió: Dear All, I'm installing the asterisk-1.6.2.24 in Centos 5.3, whenever i'm running following command ./configure I got below error: configure: *** XML documentation will not be available because the 'libxml2' development package is missing. configure: *** Please run the 'configure' script with the '--disable-xmldoc' parameter option configure: *** or install the 'libxml2' development package. I have installed already libxml2 in current os Please help me. Regards Akhilesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Skype to Asterisk
Or if you can continue to use Asterisk, you can create a gateway between FreeSwitch Skypopen Module And Asterisk. I use this solution and work very well. El 13/10/2012 04:31, Patrick Lists escribió: On 10/12/2012 11:17 PM, Philip Bennefall wrote: From what I gather, it costs extra for each channel even for direct Skype to Asterisk calls. Since my plan was to use this for business purposes, I'd need at least something like 30 channels which would be way out of my monthly budget unfortunately. If you *really* need this then have a look at FreeSWITCH which has a module for Skype calls (in/out) without the need for Skype Connect and its fees. Afaik you can use a regular Skype account and iirc even multiple Skype accounts. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send a SIP MESSAGE outside a call
Hello Tiago, if you read spanish, maybe this post can help you. http://www.voztovoice.org/?q=node/549 Regards Bakko -- _ -- 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] Less good call quality using Asterisk
Hello, I tried Asterisk Confbridge with raspberry pi without audio issue. Asterisk was compiled from sources. http://www.voztovoice.org/?q=node/553 Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Server for Asterisk
Any hint about email2fax? Thank you El 29/05/2012 17:03, Warren Selby escribió: On Tue, May 29, 2012 at 3:10 AM, Danny Dias ing.diasda...@gmail.com mailto:ing.diasda...@gmail.com wrote: Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks I've been real happy with using HylaFax+ and Iaxmodem implementations. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Trunking betweeb two Asterisk System
Hello are you using remotesecret on the trunk? regards - Original Message - From: Carlos Alvarez car...@televolve.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 23, 2012 6:08 PM Subject: Re: [asterisk-users] Trunking betweeb two Asterisk System On Thu, Feb 23, 2012 at 2:21 PM, Faraj Khasib fkha...@iconnecths.com wrote: Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Errors and other details might be helpful. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set T38 protocol
Hello, I'm trying to send a fax with sendafax aplication and receive the fax with the receiveFax aplication on the same Asterisk Server (1.8..8.2). All work fine but the PBX always use T30 protocol. Is thes a variable or setting to configure Asterisk to send and receive this fax with T38 protocol only? Thank you Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Completion
Hello, I'm trying the Call Completion system. All work fine. I still don't undesrtand how Asterisk work. Does Asterisk use sip signaling or other protocol to send notifications? On the Asterisk Wiki seems that the system is based on draft-ietf-bliss-call-completion-04 but this draft talk about SUBSCRIBE and NOTIFY. I see nothing in sip capture. Can you help me with this question? Thank's Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Completion
Thank you for your answer Kevin. Effectively, I'm using CCSS in generic mode. I hope to try the draft between two Asterisk Server soon. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
hello, yeallink T26 and T28 support OpenVPN too Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality
Hi For this scenario you can use the group and group_count functions and create a hint dialplan like this: exten = trunkname,hint,custom:trunkname When you reach the maximum number of available channels set the hint in use: Set(DEVICE_STATE(Custom:confcorso)=INUSE) To remove: Set(DEVICE_STATE(Custom:confcorso)=NOT_INUSE) After this, configure BLF on the phone top subscribe trunknumber extension Regards - Bakko -- _ -- 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] Where to find meaning of /n inLocal/6613@from-queue/n ?
http://www.voip-info.org/wiki/view/Asterisk+local+channels Regards - Original Message - From: Olivier oza_4...@yahoo.fr To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 16, 2012 7:41 AM Subject: [asterisk-users] Where to find meaning of /n inLocal/6613@from-queue/n ? Hi, Where to find meaning of /n in Local/6613@from-queue/n ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Queue option 'R'
Hello, I think this option work only with Asterisk 1.8.X On the Asterisk 1.8.X CHANGES files: * Added 'R' option to app_queue. This option stops moh and indicates ringing to the caller when an Agent's phone is ringing. This can be used to indicate to the caller that their call is about to be picked up, which is nice when one has been on hold for an extened period of time. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Client - registers but unreachable
Hello, try to configure keep alive option on Softphone if there is. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file withAuthenticateApplication
hello, you can't use authenticate for this scenario. You have to create a databse with two fields: extension and password. Then query the database with func_odbc function. There is a spanish article about this: http://www.voztovoice.org/?q=node/478 Regards - Original Message - From: virendra bhati To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 23, 2011 3:33 AM Subject: Re: [asterisk-users] How to use password file withAuthenticateApplication Hi list, I have upgrade my linux version to Asterisk 1.6.2.20. now Authenticate() function is working. But 1 question I want to add this thread.. I have 3 password in my pass.txt file. i want that only sip 2209( just example,) will come with 1234 pass and 2208 with 1235 and rest will come with 1236 password. So how I can make so ? On Tue, Nov 29, 2011 at 8:16 PM, bakko asannu...@gmail.com wrote: I use this system to authenticate my users and work fine. Asterisk: 1.6.2.20 Asterisk user: root Maybe if you active debug on the Asterisk console, you can find the error. Regards - Bakko -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suppress -- Remote UNIX connection message
Hi, maybe this parameter can help you. /etc/asterisk/asterisk.conf On the options block: hideconnect = yes Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI script that uses google's text to speech engine
Hello, when I use the Agi, sometimes not play the phrase: WARNING[30391]: file.c:650 ast_openstream_full: File ggl_U0sBo0 does not exist in any format Regards - Original Message - From: Lefteris Zafiris To: asterisk-users@lists.digium.com Sent: Wednesday, November 30, 2011 7:42 PM Subject: [asterisk-users] AGI script that uses google's text to speech engine Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first public release ca be obtained here: https://github.com/downloads/zaf/asterisk-googletts/asterisk-googletts-0.2.tar.gz To get a sample of the speech synthesis quality try this link: http://translate.google.com/translate_tts?tl=enq=this+is+a+test+for+google+text+to+speech+engine The code is still very young so suggestions, comments and bug reports are more than welcome. -- Lefteris Zafiris -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
I use this system to authenticate my users and work fine. Asterisk: 1.6.2.20 Asterisk user: root Maybe if you active debug on the Asterisk console, you can find the error. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file withAuthenticateApplication
hello, I wrote a post about this (spanish) http://www.voztovoice.org/?q=node/477 Or, if you prefere, using func_odbc http://www.voztovoice.org/?q=node/478 Regards - Bakko-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
hello, try to delete all spaces between user and password on the pass.txt Regards - Bakko-- _ -- 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] Grandstream HT503 colgado
Hola, estoy instalando un ATA HT503 de Grandstream conectado a Asterisk y todo funciona bastante bien (llamadas en entrada y salientes). El unico problema que tengo es con el colgado. Si la llamada entrante va al buzón de voz y la persona cuelga... el canal queda abierto. Será que alguien tiene o me puede indicar donde encontrar los parametros de colgado para una linea Telecom? Muchas gracias de antemano - Bakko -- _ -- 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] custom automated meeting
Hello, look at Page application Regards - Original Message - From: Thanasis thana...@asyr.hopto.org To: asterisk-users@lists.digium.com Sent: Monday, October 31, 2011 4:59 PM Subject: [asterisk-users] custom automated meeting I need your help in implementing the following scenario: A certain extension will ring two sip phones simultaneously and when one of them answers, the other keeps ringing until it answers too, and then all three (the caller and the other two) are immediately placed in a conference room (same room for all three). Can we do it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GoogleTalk Calls
Thank you Vladimir but this patch not applicable in Asterisk 1.6.2.20.
if (!strcasecmp(name, error)
-(redirect = iks_find_cdata(traversenodes, redirect))
(redirect = strstr(redirect, xmpp:))) {
redirect += 5;
ast_debug(1, redirect %s\n, redirect);
This block not exist on chan_gtalk.c
My problem is with outbound and inbound calls.
I'd like known if other users with this version have same issue.
Thank you
Regards
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[asterisk-users] GoogleTalk Calls
Hello, Is there any issue with gtalk module? Whent I try to call asterisk gtalk user nothing happens on the asterisk console. Asterisk 1.6.2.20 With Asterisk 10.0.0 beta 2 and the same configuration, works. ??? Thank you Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add PinCode on my dialplan
Hi look at option a. This option put on accountcode field the name on the left
your password file.
Regards
Enviado desde mi iPad
El 22/09/2011, a las 5:49, Malvin Rito mr...@mail.altcladding.com.ph escribió:
Hi,
I tried Authenticate where pass codes are stored on the file pass.conf and it
works.
exten = _,1,Authenticate(/etc/asterisk/pass.conf)
Since I have my CDR, I want to have a report wherein I can check which pass
code did the call. How can I achieve it?
Using authenticate through file does not replace ACCOUNT_CODE field with the
pass code entered, it only show ast_h323 under the Account_Code field.
Regards,
Malvin
On 9/21/2011 1:01 PM, Sam Govind wrote:
See core show application autheTAB
If passwords are already the same as those of voicemail.conf go for
application VMAuthenticate() - DIA generates a dial-tone which I don't think
is suitable for dialling out from users(insiders)
-= Info about application 'Authenticate' =-
[Synopsis]
Authenticate a user
[Description]
This application asks the caller to enter a given password in order to
continue
dialplan execution.
If the password begins with the '/' character, it is interpreted as a file
which contains a list of valid passwords, listed 1 password per line in the
file.
When using a database key, the value associated with the key can be anything.
Users have three attempts to authenticate before the channel is hung
up.
[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])
[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line in the
file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.
[See Also]
VMAuthenticate(), DISA()
On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito mr...@mail.altcladding.com.ph
wrote:
Thanks. ?If I want to use unique PIN for every user that dials out how can I
implement it using Authenticate app?
Regards,
Malvin
On 9/21/2011 12:42 PM, Sam Govind wrote:
DISA and DB based Auth could be an overkill.
Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application Authenticate(password[,options[,maxdigits[,prompt]]] and if
Voicemail PIN are required to be used use application
MAuthenticate([mailbox][@context][,options]
Regards,
- Sammy
On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org wrote:
Something like this should work:
exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)
exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
exten = _011.,n,Hangup
exten = _011.,n,Playback(invalid)
exten = _011.,n,Hangup
Could be cleaned up (the GotoIf isn't very descriptive about where it's
going), but it's a starting point.
On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
Hi List,
I currently have a asterisk server running used for dialing-out for IDD
but I want to Put a pincode wherein only users with the right pin code
will be allowed to dial IDD. Any sample dialplan you can suggest pls?
Thanks,
Malvin
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Re: [asterisk-users] Add PinCode on my dialplan
Or authenticate aplication. If you want use a database with a user and pin table, so each user have a pin asigned, you can look a func_odbc function. Regards - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, September 20, 2011 8:38 AM Subject: Re: [asterisk-users] Add PinCode on my dialplan That's what the DISA function is for. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito Sent: Tuesday, September 20, 2011 8:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Add PinCode on my dialplan Hi List, I currently have a asterisk server running used for dialing-out for IDD but I want to Put a pincode wherein only users with the right pin code will be allowed to dial IDD. Any sample dialplan you can suggest pls? Thanks, Malvin -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone numbers and asterisk
Hello, I think this not posible. You can use remote phonebook the phones can share. For example for yealink phone it's posible create a XML file with the phonebook and from each phone access to this list. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Integration with Android device
I think don't work with 2G network. Regards - Original Message - From: Gopal krishnan To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, August 24, 2011 4:01 PM Subject: [asterisk-users] Asterisk Integration with Android device Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreciated. Regards, Gopal -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spy just a range of extensions
Hi Alejandro, if you use 1.6.2.X look at e(ext) option With this option you can spy only the extensions you define, separate with : delimiter. Example: exten - Chanspy,1,(all,e(9000:9001:9002:9002) I don't test this option but I think work. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback while dialing out
Hi, you can configure a new music on hold, example: nano /etc/asterisk/musiconhold.conf [default1] mode=files directory=moh1 and put the audio file in this directory; then change your dialplan like: exten = 500,1,NoOp exten = 500,2,Dial(SIP/14085551234@myprovider,m(default1)) exten = 503,3,Hangup Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] display name
Hello, I use asteirsk 1.6 but i think you can set the callerid variable en asterisk 1.4 to. CALLERID(num)=test before de dial application. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which XMPP server for Jingle-enabled XMPPservice ?
I use Openfire. regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Light indicator managed by Asterisk
You can, using device_state function (I use asterisk 1.6.2.X)
Here is a example for a conference... when sombody enter to conference a
light up on my aastra phone:
exten = s,1,Set(DEVICE_STATE(Custom:confer)=INUSE)
exten = s,n,Meetme(5000)
exten = s,n,Hangup
exten = h,1,MeetMeCount(5000,users)
exten = h,2,Gotoif($[${users} = 0]?end:noend)
exten = h,3(end),Set(DEVICE_STATE(Custom:confer)=NOT_INUSE)
exten = h,4(noend),Noop(Users number = ${users})
On your subscribe context:
exten = conf,hint,custom:confer
On the phone configuration, choice BLF and asign conf to the key
Regards
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Re: [asterisk-users] Realtime - ara180
Hi, look if you have res_config_mysql.so module instaled on your asterisk. On CentOS /usr/lib/asterisk/modules Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Yealink IP Phones
Hi Giorgio, i use ftp provisioning and i think is the best solution with yealink phones. Regards - Andrea -- _ -- 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] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work
Hi, this diff solve the problem: --- include/dahdi/kernel.h 2010-08-19 20:03:25.0 +0200 +++ include/dahdi/kernel.h 2011-03-18 11:32:32.0 +0100 @@ -86,7 +86,9 @@ #endif #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,26) -#define dev_name(dev) (dev)-bus_id +#if RHEL_RELEASE_CODE RHEL_RELEASE_VERSION(5,6) +#define dev_name(dev) (dev)-bus_id +#endif #define dev_set_name(dev, format, ...) \ snprintf((dev)-bus_id, BUS_ID_SIZE, format, ## __VA_ARGS__); #endif Regards VozToVoice www.voztovoice.org i...@voztovoice.org-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read VoiceMail direct
Hi,
maybe:
exten = 8500,3,VoiceMailMain(${CALLERID(num)}@default)
Regards
- Andrea
- Original Message -
From: satish patel
To: asterisk-users
Sent: Monday, April 04, 2011 11:08 PM
Subject: [asterisk-users] Read VoiceMail direct
Hey Guy!
I want direct access of VoiceMail without asking mailbox number (Direct ask
PIN). I am using following dialplan but its still asking me Mailbox number.
Look like asterisk 1.8 doesn't support CALLERIDNUM variable.
Do you have any idea ?
exten = 8500,1,answer
exten = 8500,2,wait(1)
exten = 8500,3,voicemailmain(${CALLERIDNUM:-4}@default)
exten = 8500,4,hangup
exten = i,1,playback(invalid)
exten = i,2,hangup
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Re: [asterisk-users] Connect Asterisk to a cell phone
Hello, if you want use a Huawei 3G usb modem, take a look at chan_datacard module: http://wiki.e1550.mobi/doku.php Regards - Andrea-- _ -- 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] Error loading module 'cdr_radius.so'
Hello, you have to install radiusclient-ng http://developer.berlios.de/projects/radiusclient-ng/ Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendFAX dialplan example
Hello, you have to use a callfile http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Create a callfile, for example test.txt, on /tmp directory: Channel: SIP/providerVoIP/0317998901 Callerid: FAX WaitTime: 30 Maxretries:3 RetryTime: 300 Account: 1000 Application: SendFax Data: /var/spool/asterisk/tmp/fax.tiff Then move the file to /var/spool/asterisk/outgoing mv /tmp/test.txt /var/spool/asterisk/outgoing and look at Asterisk Console. Regards - Andrea -- _ -- 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] Unable to insert cdr-data into mysql-DB
Hi, maybe the error is on this line: sock=/tmp/mysql.sock if you use CentOS the correct line is: sock=/var/lib/mysql/mysql.sock if you use Debian/ubuntu: sock=/var/run/mysqld/mysqld.sock Regards-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH problems (asterisk 1.4.38)
Hi CLI module unload res_musiconhold.so CLI module load res_musiconhold.so or service asterisk restart Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX trunk two Asterisk
Hello, maybe is a stupid question but I'd like know if I'm doing some mistake on IAX trunk configuration. I have two Asterisk (1.6.2.14) with this configuration: A (iax.conf) register = serverb:passw...@192.168.142.246 [servera] type=friend host=dynamic trunk=yes secret=password context=phones deny=0.0.0.0/0.0.0.0 permit=192.168.142.246/255.255.255.255 qualify=yes B (iax.conf) register = servera:passw...@192.168.159.4 [serverb] type=friend host=dynamic trunk=yes secret=password context=phones deny=0.0.0.0/0.0.0.0 permit=192.168.159.4/255.255.255.255 qualify=yes The password it's the same on two server. If i use two differents passwords (one for servera and one for serverb), the trunk don't work (Call rejected No authority found) On the iax.conf general I have: calltokenoptional=0.0.0.0/0.0.0.0 Am I doing wrong something? Thank you for support Best Regards. - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme Realtime in 1.6
Hi Carlos, you have to incllude the conference options (user ad admin) in the meetme table and put schedule=yes in meetme.conf file On the dialplan just call the conference like: exten = 1557,1,Meetme(905) Regards - Bakko -- _ -- 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] Function SIP_Header not registered
Hello, I'm trying to use SIP_HEADER function on my dialplan but I receive this message (on the console): pbx.c:3367 ast_func_read: Function SIP_Header not registered Why? Thank's - Bakko -- _ -- 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] Function SIP_Header not registered
Hi Chad, thank you very much, now work... Best Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.13 IAX2 Realtime issue
Hi I have configured IAX2 realtime in Asterisk 1.6.2.13. when I cannect a client to realtime extension, always the state of extension is UNKNOW like: * Name : marco Secret : Set Context : phones Parking lot : Mailbox : 2...@default Dynamic : Yes Callnum limit: 0 Calltoken req: Auto Trunk: No Encryption : No Callerid : marco 2345 Expire : -1 ACL : No Addr-IP : XXX.XXX.XXX.XXX Port 17143 Defaddr-IP : 0.0.0.0 Port 17143 Username : marco Codecs : 0xc (ulaw|alaw) Codec Order : (alaw|ulaw) Status : UNKNOWN Qualify : every 6ms when OK, every 1ms when UNREACHABLE (sample smoothing Off) I tried with zoiper and DIAX softhone without success. Any hint? Thank's - bakko -- _ -- 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] Festival
Hi, wich version of Asterisk? If is 1.6.2.13, there is a open issue becouse not work https://issues.asterisk.org/view.php?id=17995 R.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addons for Asterisk 1.8?
The addons are in the same package. Regards - Original Message - From: Carlos Chavez cur...@telecomabmex.com To: Asterisk asterisk-users@lists.digium.com Sent: Monday, November 08, 2010 4:43 PM Subject: [asterisk-users] Addons for Asterisk 1.8? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_ais Error
Hi,
I'm trying distributed events with Openais but don't work.
I made the test with two asterisk box in the same LAN
box A: 192.168.142.246 asterisk 1.6.2.13
BoxB: 192.168.142.248 asterisk 1.8.0
openais.conf:
# Please read the openais.conf.5 manual page
totem {
version: 2
secauth: off
threads: 0
consensus: 4800
interface {
ringnumber: 0
bindnetaddr: 192.168.142.0
mcastaddr: 226.94.1.1
mcastport: 5405
}
}
logging {
to_stderr: yes
debug: on
timestamp: on
to_file: yes
to_syslog: no
syslog_facility: daemon
logfile: /var/log/openais.log
}
amf {
mode: disabled
}
When I load res_ais.so module, the pbx crash (boths)
Some time not crash but no clusters members are present.
What I'm doing wrong?
Thank's
Regards
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Re: [asterisk-users] res_ais Error
Hi, after some test the system don't crash but no members: CLI ais show clm members = === Cluster Members = = === === - === Node Name: === == ID: 0x0 === == Address: === == Member: No === - === = In the openais.log i can see the connection but I don't know why not persistent: Nov 5 23:36:59.361077 [ipc.c:0799] connection received from libais client 5. Nov 5 23:36:59.363573 [ipc.c:0799] connection received from libais client 6. Nov 5 23:37:04.707123 [TOTEM] The consensus timeout expired. Nov 5 23:37:04.707207 [TOTEM] entering GATHER state from 3. Nov 5 23:37:10.526431 [TOTEM] The consensus timeout expired. Nov 5 23:37:10.526506 [TOTEM] entering GATHER state from 3. Nov 5 23:37:16.345764 [TOTEM] The consensus timeout expired. Nov 5 23:37:16.345829 [TOTEM] entering GATHER state from 3. Nov 5 23:37:22.165079 [TOTEM] The consensus timeout expired. Nov 5 23:37:22.165181 [TOTEM] entering GATHER state from 3. Nov 5 23:37:27.984413 [TOTEM] The consensus timeout expired. Nov 5 23:37:27.984534 [TOTEM] entering GATHER state from 3. Nov 5 23:37:33.803701 [TOTEM] The consensus timeout expired. Nov 5 23:37:33.803799 [TOTEM] entering GATHER state from 3. Nov 5 23:37:39.621992 [TOTEM] The consensus timeout expired. Nov 5 23:37:39.622067 [TOTEM] entering GATHER state from 3. Nov 5 23:37:45.441295 [TOTEM] The consensus timeout expired. Nov 5 23:37:45.441376 [TOTEM] entering GATHER state from 3. Any idea? Regards - bakko -- _ -- 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] being bombarded with SIP packets
Fail2Ban Regards - Original Message - From: Per Jessen p...@computer.org To: asterisk-users@lists.digium.com Sent: Thursday, October 28, 2010 2:41 AM Subject: [asterisk-users] being bombarded with SIP packets Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] generic_odbc and ltdl are not available to enableODBC support
Hi, are you installed unixodbc-dev? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto provisioning from public server
Hello, many SIP phones offer you the possibility to provisioning them over a FTP connection (with username and password). Regards - Bakko -- _ -- 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] Error with Connecting Two Asterisk BOX with IAX
Yhank you very much Giorgio, now work with the general option: calltokenoptional=0.0.0.0/0.0.0.0 Regards - Bakko -- _ -- 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] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi Bruce, can you show agent login/logoff diaplan? Maybe there is a solution but i have to know how yours agents login/logoff. Regards - Bakko -- _ -- 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] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi Bruce, with this configuration you can`t control the state of agent. Sorry Regards-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Calls Rejection Reasons
Yes, look at DIALSTATUS variable that Asterisk set when use DIAL Application. Regards - Bakko-- _ -- 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] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hello, you can't utilice the same butons to know the state of the agent but you can configure the LEDs in the opposite position (4,5,6) in the dialplan just before the command to login to the queue put this line (for english queue): exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=INUSE) for spanish queue exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=INUSE) in the dialplan part relative to agent logoff (english) exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=NOT_INUSE) spanish exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=NOT_INUSE) then on the Aastra 6755i web page (on the Programmable Keys menu): keytypevalueline 4BLFagentenglobal 5BLFagentesglobal Now each time the agent login to english queue the 4 key LED switch to red. The same with key 5 LED Please try and give us a feedback Regards - Bakko -- _ -- 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] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi, you can use 4 for login/logoff (english and spanish) and two for online/offline The procedure is the same. Regards - Bakko-- _ -- 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] IAX2 works one direction, but not the other...
Hi Paul, I spent two days to conect two Asterisk BOX (1.6.2.13) with IAX with username and password. Only when i changed secret with remotesecret the connection work. Maybe you can try the same configuration to confirm this behaviour Regards - Bakko -- _ -- 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] IAX2 works one direction, but not the other...
Hi Paul, maybe there is some think wrong on iax. if I set remotesecret on IAX2 extension the call from Server A to Server B work but not authenticated and host is set to dynamic (normaly if is a IP authentication on host parametre I put the IP) If I set secret on two box and both are registered without errors, when i call from box A or from box B to box A, always I receive this error No auhthority found. Thi behaviour only happens if the Asterisk version onTwo box is 1.6.2.13. If Asterisk version on Box A is 1.4.X and Asterisk version on Box B is 1.6.2.13 (with the same configuration) work fine. Why? Thank you in advance. Regards - Bakko -- _ -- 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] Error with Connecting Two Asterisk BOX with IAX
Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register = coiax:pa...@69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:456960 Registered iax2 peers smiax69.164.207.166 (D) 255.255.255.255 4569 (T) OK (3 ms) Asterisk B: register = smiax:pa...@69.164.197.105 [coiax] type=friend host=dynamic trunk=yes secret=pass1 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.197.105/255.255.255.255 qualify=yes Console iax2 registry 69.164.197.105:4569 N smiax 69.164.207.166:456960 Registered iax2 peers coiax69.164.197.105 (D) 255.255.255.255 4569 (T) OK (3 ms) When I try to call from Asterisk A to Asterisk B I receive this error Asterisk A WARNING[28759]: chan_iax2.c:10287 socket_process: Call rejected by 69.164.207.166: No authority found AsteriskB NOTICE[26563]: chan_iax2.c:10522 socket_process: Host 69.164.197.105 failed to authenticate as coiax What's wrong? Thank you in advance. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
Hi Flavio, try with this funtion before the line with the english meetme application Set(CHANNEL(language)=en) and Set(CHANNEL(language)=pr) before the line with the portugues meetme application Regards - Bakko-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
Hi Flavio is: [conference] exten = 1001,3,Set(CHANNEL(language)=pt_BR) exten = 1001,4,MeetMe(1001,ipdM) exten = 1001,5,Playback(vm-goodbye) exten = 1001,6,Hangup Regards - Bakko-- _ -- 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 CDR Radius error
Hello, I'm trying to configure Asterisk with Radius cdr support. Asterisk version 1.6.2.13 Server Radius: Freeradius version 1.X Radius client: radiusclient-ng version 0.5.5 With the Asterisk core debug on 1 when a call terminate, on the console appear this error: Unable to create RADIUS record. CDR not recorded! My cdr.conf is: [radius] usegmtime=yes; log date/time in GMT loguniqueid=yes ; log uniqueid loguserfield=yes ; log user field radiuscfg=/etc/radiusclient-ng/radiusclient.conf When I load the cdr_radius module no error appear. Any suggestion? Regards - Andrea -- _ -- 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] Setting up realtime config.
Hi Mike, Which is the real name for this peer? If you want look the configuration peer on Asterisk console try: CLI sip show peer accountname load To register to this account on Ekiga... accountname is the name of the extensions you have to configure. BR - Andrea -- _ -- 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] Setting up realtime config.
Hi Mike, Which is the real name for this peer? If you want look the configuration peer on Asterisk console try: CLI sip show peer accountname load To register to this account on Ekiga... accountname is the name of the extensions you have to configure. BR - Andrea -- _ -- 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 CDR Radius error
Hi, Have you got a dictionary file with the attributes for asterisk? Yes, my radiusclient-ng dictionary include dictionary.digium BR -- _ -- 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
