Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
On Wed, 10 Feb 2010, Muro, Sam wrote: [snip] I have the agi scripts not as ivr but to help populate the required information into mysql db. Probably here is where the problem lies i have to connect and disconnect to mysql each time a call is made or a specific menu is selected Here is the script * #!/usr/bin/perl -w [snip] You can execute xxx AGIs written in a compiled language like C in the time it takes to load the Perl interpreter and parse your script. [snip] # Trying to resolve memory leak should it happen delete($ARGV[0]); delete($ARGV[1]); delete($ARGV[2]); delete($ARGV[3]); delete($ARGV[4]); delete($ARGV[5]); delete($ARGV[6]); delete($ARGV[7]); Not part of your issue, but any memory leakage in a process (which is what Asterisk creates to execute your AGI) is automagically cleaned up when the process is terminated. Since all your AGI does is take a bunch of channel variables (I'm assuming a little bit here) and stuff them into your database, this would be simple (but ugly) to code directly into the dialplan. My preference would be to keep the database cruft in an AGI written in C. The performance should be an order of magnitude or two better and you can keep your dialplan clean and maintainable. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
- Sam Muro resea...@businesstz.com wrote:
Hi Team
Can someone advice me on how i can lower the load average on my
asterisk server?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1
2 X TE412P Digium cards on ISDN PRI
Im using the system as an IVR without any transcoding or bridging
**
top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75,
62.55,
55.75
Hi Sam!
Hello Steve!
Are there any side-effects from the high load average? The system
doesn't seem to be CPU or disk bound from the look of the CPU stats.
System %age is
high by way - software echo cancellaton?, and Asterisk is using a
lot of
cpu
which isn't suprising.
Yes. Audio quality issues. I have enabled the hardware echo
cancellation
and configured
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
I'm guessing you are running 8 spans and 200+ calls into your IVR?
You are correct. 8 span which process up to 240 calls at pick time
If the system is actually performing fine then I'd just say that
there
is something about the Asterisk threads that makes them look runnable
and that
accounts for the high load average. Is the IVR an agi or fastagi
or
what? -
I have the agi scripts not as ivr but to help populate the required
information into mysql db. Probably here is where the problem lies i
have
to connect and disconnect to mysql each time a call is made or a
specific
menu is selected
Here is the script
*
#!/usr/bin/perl -w
use strict;
use DBI();
use Scalar::Util qw/weaken/;
my $cdr_log_file = /var/log/asterisk/ivr_log;
my $mysql_host = cdr01;
my $mysql_db = ivrcdrdb;
my $mysql_table = tbl_ivrcdr_details;
my $mysql_user = ivruser;
my $mysql_pwd = a09876a;
my $sth;
my $data0= $ARGV[0];
my $data1= $ARGV[1];
my $data2= $ARGV[2];
my $data3= $ARGV[3];
my $data4= $ARGV[4];
my $data5= $ARGV[5];
my $data6= $ARGV[6];
my $data7= $ARGV[7];
# Connect to database
# print Connecting to database...\n\n;
my $dbh =
DBI-connect(DBI:mysql:database=$mysql_db;host=$mysql_host,$mysql_user,$mysql_pwd,{'RaiseError'
= 1});
my $insert_str = insert into $mysql_table (calldate, language, src,
duration, accountcode, uniqueid, currentmenu, nextmenu) values
(\$data0\, \$data1\, \$data2\, \$data3\, \$data4\,
\$data5\,
\$data6\, \$data7\);\n;
$sth = $dbh-prepare($insert_str);
$sth-execute();
# print \n\nOK.\n;
$sth-finish();
$dbh-disconnect();
# Trying to resolve memory leak should it happen
delete($ARGV[0]);
delete($ARGV[1]);
delete($ARGV[2]);
delete($ARGV[3]);
delete($ARGV[4]);
delete($ARGV[5]);
delete($ARGV[6]);
delete($ARGV[7]);
exit;
*
the code path may have a spinlock logic to it that means that
many
threads
are runnable but when scheduled just go back to sleep. That would
account for high load average with lots of spare CPU. If that's what
is
happening then I wouldn't worry much more about it.
Regards,
Steve
Regards
Sam
Perhaps change the PASSWORD as well!
--
Thanks, Phil
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[asterisk-users] asterisk and mysql connection
Hell list. I wanna use mysql for storing user’s ID or etc. If user call to other, asterisk have to search number in mysql. Are there document about setting asterisk and mysql? THX Kim -- _ -- 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 and mysql connection
- 김무성 ki...@infosec.co.kr wrote: Hell list. I wanna use mysql for storing user’s ID or etc. If user call to other, asterisk have to search number in mysql. Are there document about setting asterisk and mysql? http://www.voip-info.org/wiki/view/Asterisk+RealTime -- Thanks, Phil -- _ -- 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] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
snip
You are correct. 8 span which process up to 240 calls at pick time
If the system is actually performing fine then I'd just say that there
is something about the Asterisk threads that makes them look runnable
and that
accounts for the high load average. Is the IVR an agi or fastagi or
what? -
I have the agi scripts not as ivr but to help populate the required
information into mysql db. Probably here is where the problem lies i have
to connect and disconnect to mysql each time a call is made or a specific
menu is selected
Here is the script
*
#!/usr/bin/perl -w
use strict;
use DBI();
use Scalar::Util qw/weaken/;
my $cdr_log_file = /var/log/asterisk/ivr_log;
my $mysql_host = cdr01;
my $mysql_db = ivrcdrdb;
my $mysql_table = tbl_ivrcdr_details;
my $mysql_user = ivruser;
my $mysql_pwd = a09876a;
my $sth;
my $data0= $ARGV[0];
my $data1= $ARGV[1];
my $data2= $ARGV[2];
my $data3= $ARGV[3];
my $data4= $ARGV[4];
my $data5= $ARGV[5];
my $data6= $ARGV[6];
my $data7= $ARGV[7];
# Connect to database
# print Connecting to database...\n\n;
my $dbh =
DBI-connect(DBI:mysql:database=$mysql_db;host=$mysql_host,$mysql_user,$mysql_pwd,{'RaiseError'
= 1});
my $insert_str = insert into $mysql_table (calldate, language, src,
duration, accountcode, uniqueid, currentmenu, nextmenu) values
(\$data0\, \$data1\, \$data2\, \$data3\, \$data4\, \$data5\,
\$data6\, \$data7\);\n;
$sth = $dbh-prepare($insert_str);
$sth-execute();
# print \n\nOK.\n;
$sth-finish();
$dbh-disconnect();
# Trying to resolve memory leak should it happen
delete($ARGV[0]);
delete($ARGV[1]);
delete($ARGV[2]);
delete($ARGV[3]);
delete($ARGV[4]);
delete($ARGV[5]);
delete($ARGV[6]);
delete($ARGV[7]);
exit;
*
the code path may have a spinlock logic to it that means that many
threads
are runnable but when scheduled just go back to sleep. That would
account for high load average with lots of spare CPU. If that's what is
happening then I wouldn't worry much more about it.
Regards,
Steve
Regards
Sam
If I were you, and I am not and never will be, I would move over to
fastagi and offload all that Perl and database stuff off to a
designated server just to handle that stuff.
I have had the EXACT same problem and that is how it was fixed,
fastagi running to a Windows box that had a process developed (written
in C something) by the M$ developers to hit the M$SQL databases.
We were also doing a ton of things with the AMI which we figured out
how to do the same end result without banging on the AMI, such as
using call files rather than AMI to originate a call.
Load avg dropped to one or under if I remember correctly.
Thanks,
Steve Totaro
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Re: [asterisk-users] asterisk and mysql connection
김무성 wrote: Hell list. I wanna use mysql for storing user’s ID or etc. If user call to other, asterisk have to search number in mysql. Are there document about setting asterisk and mysql? THX Kim Hi You need to read up on Asterisk Realtime http://www.voip-info.org/wiki/view/Asterisk+RealTime Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
On Wed, Feb 10, 2010 at 10:12:55AM +0300, Muro, Sam wrote: Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Tasks: 149 total, 1 running, 148 sleeping, 0 stopped, 0 zombie Cpu0 : 10.3%us, 32.0%sy, 0.0%ni, 57.3%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu1 : 10.6%us, 34.6%sy, 0.0%ni, 54.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu2 : 13.3%us, 36.5%sy, 0.0%ni, 49.8%id, 0.0%wa, 0.0%hi, 0.3%si, 0.0%st Cpu3 : 8.6%us, 39.5%sy, 0.0%ni, 51.8%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu4 : 7.3%us, 38.0%sy, 0.0%ni, 54.7%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu5 : 17.9%us, 37.5%sy, 0.0%ni, 44.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu6 : 13.3%us, 37.2%sy, 0.0%ni, 49.5%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu7 : 12.7%us, 37.3%sy, 0.0%ni, 50.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st System is fairly loaded, but there's still plenty of idle CPU cycles. If we were in a storm of CPU-intensive processes, we would have expected many more running processes. Right now we have none (the single process is 'top' itself). Mem: 3961100k total, 3837920k used, 123180k free, 108944k buffers Swap: 779144k total, 56k used, 779088k free, 3602540k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 683 root 15 0 97968 36m 5616 S 307.7 0.9 41457:34 asterisk 17176 root 15 0 2196 1052 800 R 0.7 0.0 0:00.32 top 1 root 15 0 2064 592 512 S 0.0 0.0 0:13.96 init 2 root RT -5 000 S 0.0 0.0 5:27.80 migration/0 3 Processes seem to be sorted by size. You should have pressed 'p' to go back to sorting by CPU. Now we don't even see the worst offenders. Tried option 'p' but doesnt seems to exist. Centos 5.3 kernel 2.6.18-128 Sorry: shift-p (and shift-m to sort by memory). Another handy switch: shift-h to toggle the display of different threads of the same process separately. root 34 19 000 S 0.0 0.0 0:00.11 ksoftirqd/0 4 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/0 5 root RT -5 000 S 0.0 0.0 1:07.67 migration/1 6 root 34 19 000 S 0.0 0.0 0:00.09 ksoftirqd/1 7 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/1 8 root RT -5 000 S 0.0 0.0 1:16.92 migration/2 9 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/2 10 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/2 11 root RT -5 000 S 0.0 0.0 1:34.54 migration/3 12 root 34 19 000 S 0.0 0.0 0:00.15 ksoftirqd/3 13 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/3 14 root RT -5 000 S 0.0 0.0 0:54.66 migration/4 15 root 34 19 000 S 0.0 0.0 0:00.01 ksoftirqd/4 16 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/4 17 root RT -5 000 S 0.0 0.0 1:39.64 migration/5 18 root 39 19 000 S 0.0 0.0 0:00.21 ksoftirqd/5 19 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/5 20 root RT -5 000 S 0.0 0.0 1:06.27 migration/6 21 root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/6 22 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/6 23 root RT -5 000 S 0.0 0.0 1:23.24 migration/7 24 root 34 19 000 S 0.0 0.0 0:00.17 ksoftirqd/7 25 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/7 26 root 10 -5 000 S 0.0 0.0 0:25.70 events/0 27 root 10 -5 000 S 0.0 0.0 0:37.83 events/1 28 root 10 -5 000 S 0.0 0.0 0:15.67 events/2 29 root 10 -5 000 S 0.0 0.0 0:40.36 events/3 30 root 10 -5 000 S 0.0 0.0 0:16.45 events/4 Those are all kernel threads rather than real processes. So I suspect one of two things: 1. You're right after such a storm. The load average will decreases sharply. What do you mean Trafrir Its obvious that the effect increases with increase number of active channels. e.g. @channels=90, load average = 4 but @channels =235, load average= 60+ Each Asterisk channel has a separate thread. The thing that looked odd was that there were no processes (actually: threads. The Linux scheduler schdules threads). The load average is the average length of the running queue over a certain period of time (three numbers: first one is over a period of a minute,
[asterisk-users] PMS (SDMR, ...) support in Asterisk
Hello, In this list archives, you can find here and there threads related to Property Management System support in Asterisk. Google shows this doc (http://www.mitel.com/resources/guide_8922_misn.pdf) which gives an interesting overview of this topic. 1. Is this Station Message Detail Recording widely used between PBXs and call accounting software ? 2. Are you aware of a protocol allowing guest telephones provisionning (when a guest checks in, its telephone is activated and personnalized) ? 3. What amount of work would it take to develop an SMDR interface in Asterisk. It seems several people have done (or planned to) this and a return of experience would be appreciated. 4. What is the most widely used call accounting software in Hotel sector ? Best regards -- _ -- 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] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
On Wed, Feb 10, 2010 at 04:26:48AM -0500, Steve Totaro wrote: [snip] my $data0= $ARGV[0]; my $data1= $ARGV[1]; my $data2= $ARGV[2]; my $data3= $ARGV[3]; my $data4= $ARGV[4]; my $data5= $ARGV[5]; my $data6= $ARGV[6]; my $data7= $ARGV[7]; [snip] my $insert_str = insert into $mysql_table (calldate, language, src, duration, accountcode, uniqueid, currentmenu, nextmenu) values (\$data0\, \$data1\, \$data2\, \$data3\, \$data4\, \$data5\, \$data6\, \$data7\);\n; $sth = $dbh-prepare($insert_str); $sth-execute(); If I were you, and I am not and never will be, I would move over to fastagi and offload all that Perl and database stuff off to a designated server just to handle that stuff. Or, in the case of such a simple AGI, use the MySQL app from addons. Alternatively, can the CDR mechnism be (ab?)used to record this information? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] forward incomming line to modem
On Wed, Feb 10, 2010 at 07:52:06AM +0100, randall wrote: hi All, its probably very simple but i can't find the way to it. i have some b410p cards and use them to connect to ISDN2, this works OK for calling but i need to have 1 line to be send to the fax machine. BRI fax machine? the fax machine is a modem connected on another machine with hylafax. as far as i can figure out i need to set 1 of the slots, the one leading to the fax, in the b410p in NT mode by setting the jumpers in the opposite direction as the default. Not sure how to proceed after that but dahdi keeps showing this port as TE mode. Look in your logs and you'll find the message How cool would it be if someone implemented this mode! For now, sucks for you. Use bri_net instead of bri_net_ptmp . anybody has a pointer, i'm pretty much stuck and i guess its simply that i'm not feeding google the right buzz word. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
snip
You are correct. 8 span which process up to 240 calls at pick time
If the system is actually performing fine then I'd just say that there
is something about the Asterisk threads that makes them look runnable
and that
accounts for the high load average. ?Is the IVR an agi or fastagi or
what? -
I have the agi scripts not as ivr but to help populate the required
information into mysql db. Probably here is where the problem lies i
have
to connect and disconnect to mysql each time a call is made or a
specific
menu is selected
Here is the script
*
#!/usr/bin/perl -w
use strict;
use DBI();
use Scalar::Util qw/weaken/;
my $cdr_log_file = /var/log/asterisk/ivr_log;
my $mysql_host = cdr01;
my $mysql_db = ivrcdrdb;
my $mysql_table = tbl_ivrcdr_details;
my $mysql_user = ivruser;
my $mysql_pwd = a09876a;
my $sth;
my $data0= $ARGV[0];
my $data1= $ARGV[1];
my $data2= $ARGV[2];
my $data3= $ARGV[3];
my $data4= $ARGV[4];
my $data5= $ARGV[5];
my $data6= $ARGV[6];
my $data7= $ARGV[7];
# Connect to database
# print Connecting to database...\n\n;
my $dbh =
DBI-connect(DBI:mysql:database=$mysql_db;host=$mysql_host,$mysql_user,
$mysql_pwd,{'RaiseError'
= 1});
my $insert_str = insert into $mysql_table (calldate, language, src,
duration, accountcode, uniqueid, currentmenu, nextmenu) values
(\$data0\, \$data1\, \$data2\, \$data3\, ?\$data4\,
\$data5\,
\$data6\, \$data7\);\n;
? ? ? $sth = $dbh-prepare($insert_str);
? ? ? $sth-execute();
# print \n\nOK.\n;
$sth-finish();
$dbh-disconnect();
# Trying to resolve memory leak should it happen
delete($ARGV[0]);
delete($ARGV[1]);
delete($ARGV[2]);
delete($ARGV[3]);
delete($ARGV[4]);
delete($ARGV[5]);
delete($ARGV[6]);
delete($ARGV[7]);
exit;
*
the code path may have a spinlock logic to it that means that many
threads
are runnable but when scheduled just go back to sleep. ?That would
account for high load average with lots of spare CPU. ?If that's what is
happening then I wouldn't worry much more about it.
Regards,
Steve
Regards
Sam
If I were you, and I am not and never will be, I would move over to
fastagi and offload all that Perl and database stuff off to a
designated server just to handle that stuff.
I have had the EXACT same problem and that is how it was fixed,
fastagi running to a Windows box that had a process developed (written
in C something) by the M$ developers to hit the M$SQL databases.
We were also doing a ton of things with the AMI which we figured out
how to do the same end result without banging on the AMI, such as
using call files rather than AMI to originate a call.
Load avg dropped to one or under if I remember correctly.
Thanks,
Steve Totaro
Thank you Steve for your recommendation. Ofcoz i have separate server that
is hosting the db and i will consider doing fastagi and see it it will help
@Phil. The credintials displayed there are dummy, so don't worry unless you
mean something else
@Steve Edward. Can you share your C agi codes? I presume what you want me to
do is rewrite the script in C and use it as compiled binary
@Tzafrir. How about this
[ivr4 ~]# ps aux | grep D
USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND
root 1975 0.0 0.0 3920 688 pts/4S+ 13:17 0:00 grep D
root 3413 0.0 0.0 1832 576 ?Ss2009 80:58
/usr/sbin/mDNSResponder -b -f /etc/services_mDNS
I have killed that process but no changes
@All, looks like the conclusion has been made that this is to do with AGI.
Let me address it and see how it reacts. I shall feedback
Thanks
Sam
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Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
On Wed, Feb 10, 2010 at 01:23:01PM +0300, RESEARCH wrote: @Steve Edward. Can you share your C agi codes? I presume what you want me to do is rewrite the script in C and use it as compiled binary Yes. But then again, for such a simple call (a single INSERT) you can use a MySQL() from the dialplan. And this is also basically the same as adding a CDR record. Which is why I guess you can probably use the existing CDR code. @Tzafrir. How about this [ivr4 ~]# ps aux | grep D USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND root 1975 0.0 0.0 3920 688 pts/4S+ 13:17 0:00 grep D root 3413 0.0 0.0 1832 576 ?Ss2009 80:58 /usr/sbin/mDNSResponder -b -f /etc/services_mDNS So no processes are currently in state 'D' (the two processes here are grep itself, and an unrelated daemon, both in state 'S', which is the normal 'Sleeping'). (So your issue was merely the fact the the CPU was flooded with work, and not some processes hung in uninterruptable sleep). I have killed that process but no changes * Killing a process in state D is generally pointless (it is in an uniteruuptable system call: not even kill -9 would get it). * That process was not in state D anyway :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nat Issue - is this Draytek || Asterisk?
I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT Routed' on a single device - so my Asterisk sits on a Public IP, and I have a number of SIP clients on a LAN being natted. If I open a single client on the LAN, it all works as expected. However, if another machine on the LAN opens a client no client will work. Attempting to call anything like Voicemail fails and after a short while Asterisk starts scrolling: [Feb 10 11:10:31] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-...@192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Feb 10 11:10:31] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-...@192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Feb 10 11:10:32] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-...@192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. The only way to get service back is kill all other clients on the LAN and restart the router. Naturally I'm questioning the router, but the fly in the ointment is that it worked before I upgraded from 1.6.1 to 1.6.2 - which makes me think that it could be Asterisk itself. I'm starting to wonder if there is an issue in the Asterisk NAT code as I'm also seeing some 'stale nonce received' relating to the LAN IP of the second client after I disconnect it. I'm struggling to work out how can I debug this effectively and would appreciate some guidance here. -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
Since upgrading from 1.6.1 to 1.6.2 I get this error on boot: Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory Or if I try to connect to Asterisk: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) If I manually create /var/run/asterisk/ and restart Asterisk I can connect to it, but if the server is rebooted /var/run/asterisk/ disappears and warning comes back. I could doctor the init.d script to overcome this, but I'm not sure it's the right thing to do. Can anyone explain the best way for me to get over this? -- _ -- 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] forward incomming line to modem
On Wed, 2010-02-10 at 12:02 +0200, Tzafrir Cohen wrote: On Wed, Feb 10, 2010 at 07:52:06AM +0100, randall wrote: hi All, its probably very simple but i can't find the way to it. i have some b410p cards and use them to connect to ISDN2, this works OK for calling but i need to have 1 line to be send to the fax machine. BRI fax machine? no, its an external analog modem the fax machine is a modem connected on another machine with hylafax. as far as i can figure out i need to set 1 of the slots, the one leading to the fax, in the b410p in NT mode by setting the jumpers in the opposite direction as the default. Not sure how to proceed after that but dahdi keeps showing this port as TE mode. right, followed the wrong manual that told me to use the 2 switches instead of the jumper. i feel stupd. at least it shows NT now. Look in your logs and you'll find the message How cool would it be if someone implemented this mode! For now, sucks for you. pfff, is it just me or is the readability of the asterisk project not always up to par? , wouldn't want to say it sucked. Use bri_net instead of bri_net_ptmp . i've got bri_net as created by dahdi_genconf on the line connected to the modem, the incoming lines are bri_cpe. the bri description doesn't make sense to me for connecting to an analog device. whats next? all the info i can find on this subject relates to hylafax being installed on the same server. have set faxdetect=incoming faxdetect=outgoing faxdetect=yes so i guess it would detect an incoming fax automagically. what kind of extension do you need to pass the signal to? anybody has a pointer, i'm pretty much stuck and i guess its simply that i'm not feeding google the right buzz word. -- _ -- 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] problems with creating a call
Hello, I installed Asterisk in a linonde cloud debian 5, and i'm trying to create a first call but when i try to set up the call i see the following message: -- Called 1...@100 -- Now forwarding SIP/105-0008 to 'Local/1...@default' (thanks to SIP/100-0009) -- Executing [...@default:1] Dial(Local/1...@default-c2a9;2, SIP/1...@100) in new stack [Feb 10 13:31:25] WARNING[3639]: app_dial.c:1712 dial_exec_full: Skipping dialing interface 'SIP/1...@100' again since it has already been dialed i'm calling from 105 to 100 (100 is registred at another domain, defined in sip.conf that's why there is an 1...@100) I hope anybody has any input on this because i'm lost :-) never had this.. it's just a simple dial.. Thanks, Peter -- _ -- 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] problems with creating a call
Peter den Hartog wrote: Hello, I installed Asterisk in a linonde cloud debian 5, and i'm trying to create a first call but when i try to set up the call i see the following message: -- Called 1...@100 -- Now forwarding SIP/105-0008 to 'Local/1...@default' (thanks to SIP/100-0009) -- Executing [...@default:1] Dial(Local/1...@default-c2a9;2, SIP/1...@100) in new stack [Feb 10 13:31:25] WARNING[3639]: app_dial.c:1712 dial_exec_full: Skipping dialing interface 'SIP/1...@100' again since it has already been dialed i'm calling from 105 to 100 (100 is registred at another domain, defined in sip.conf that's why there is an 1...@100) The device at SIP/100 sent a redirect (forward) message back to Asterisk suggesting that the call be sent to extension '100'. Asterisk refuses to call that device again because it's already been called in that particular instance of Dial and doing so would just result in an infinite loop. You need to figure out why the device at SIP/100 told Asterisk to forward the call when you were expecting it to just accept it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
Brian, It could be that the ownership/permissions on the directory are not correct. Are you running asterisk as asterisk:asterisk or root:root? Here is an article that lists the directories and what the ownership/permissions on each one should be: http://www.voip-info.org/wiki/view/Asterisk+non-root On Wed, Feb 10, 2010 at 11:57:44AM +, Brian wrote: Since upgrading from 1.6.1 to 1.6.2 I get this error on boot: Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory Or if I try to connect to Asterisk: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) If I manually create /var/run/asterisk/ and restart Asterisk I can connect to it, but if the server is rebooted /var/run/asterisk/ disappears and warning comes back. I could doctor the init.d script to overcome this, but I'm not sure it's the right thing to do. Can anyone explain the best way for me to get over this? -- _ -- 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 -- _ -- 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] Muted calls occasionally dropping after 30 seconds
Hi I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. The rtptimeout and rtpholdtimeout are both set to 0 on a global level and for the sip extension the sip table row has NULL in both columns. I've tried playing with those 2 values, both on a global and sip extension level but regardless to what they are set to, if the call gets disconnected it is always 30 seconds after the mute button is pressed. But like I said before, this does not happen every time the mute button is pressed. I managed to recreate the phenomenon one one of our test servers so I could be certain that there was nothing else going on at the time. The call path when recreating this on our test platform was My Mobile - number/SIP provider - out asterisk server - SIP extension Has anyone else ever experienced anything like this? It's really got me rather frustrated! Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
On Wed, 2010-02-10 at 08:54 -0500, Ken Leland III wrote: Brian, It could be that the ownership/permissions on the directory are not correct. Are you running asterisk as asterisk:asterisk or root:root? Here is an article that lists the directories and what the ownership/permissions on each one should be: http://www.voip-info.org/wiki/view/Asterisk+non-root Thanks for that, but no - it runs as root, and root can create/access /var/run without any issues. I'm boggled... -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
It seems to me that the restart is creating asterisk.pid in the wrong place. Try this - - find /|grep asterisk.pid This will tell you where the mislocated pid is being created and you can adjust the script accordingly. -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Sent: Wednesday, February 10, 2010 8:15 AM To: Ken Leland III Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory On Wed, 2010-02-10 at 08:54 -0500, Ken Leland III wrote: Brian, It could be that the ownership/permissions on the directory are not correct. Are you running asterisk as asterisk:asterisk or root:root? Here is an article that lists the directories and what the ownership/permissions on each one should be: http://www.voip-info.org/wiki/view/Asterisk+non-root Thanks for that, but no - it runs as root, and root can create/access /var/run without any issues. I'm boggled... -- _ -- 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 -- _ -- 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] Optimization of call from server 1 to 2 and then back to 1
Hi All, suppose this call flow: there are two Asterisk servers, they are connected through a IAX2 trunk. The users use SIP. The user A on the Asterisk server 1 calls the user B on the Asterisk server 2. They talk for a while and then the user B does an attendant transfer to the user C on the Asterisk server 1. Question: is it possible to optimize the voice flow or the music on hold flow so that it is done inside the Asterisk server 1 instead of forward and back: from server 1 to 2 and then back to 1 ? Thanks for your attention and for supporting, have a nice day. Mike -- _ -- 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] Optimization of call from server 1 to 2 and thenback to 1
Difficult to say since you don't say if you are on 1.2, 1.4 or 1.6, but my WAG would be that the IAX connection takes this out Asterisk 1's hands. The attendant transfer never breaks the IAX connection; it actually creates an extra IAX connection to let A talk to C like this: Original call A -- IAX -- B B -- IAX -- C = A -- IAX -- IAX -- C You should be able to verify this with a core show channels during the two legs. At any rate, MOH is controlled by the holding party, so when A puts B or C on hold, Asterisk 1 is controlling; B - Asterisk 2; C - Asterisk 2 via IAX; Go ahead, shoot me down if I'm wrong; just an educated WAG -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mancyb...@gmail.com Sent: Wednesday, February 10, 2010 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Optimization of call from server 1 to 2 and thenback to 1 Hi All, suppose this call flow: there are two Asterisk servers, they are connected through a IAX2 trunk. The users use SIP. The user A on the Asterisk server 1 calls the user B on the Asterisk server 2. They talk for a while and then the user B does an attendant transfer to the user C on the Asterisk server 1. Question: is it possible to optimize the voice flow or the music on hold flow so that it is done inside the Asterisk server 1 instead of forward and back: from server 1 to 2 and then back to 1 ? Thanks for your attention and for supporting, have a nice day. Mike -- _ -- 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 -- _ -- 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] problems with creating a call
hehe i figured it out.. it was really stupid :) i use opensips as an sip proxy, and i configured opensips to only react on packages from local ip's.. asterisk was sending to an external ip and that way i created my own little loop :), changed in sip.conf all the hosts to the internal ip of opensips and it worked.. thanks for the input tho :)! Peter On Wed, Feb 10, 2010 at 2:44 PM, Kevin P. Fleming kpflem...@digium.comwrote: Peter den Hartog wrote: Hello, I installed Asterisk in a linonde cloud debian 5, and i'm trying to create a first call but when i try to set up the call i see the following message: -- Called 1...@100 -- Now forwarding SIP/105-0008 to 'Local/1...@default' (thanks to SIP/100-0009) -- Executing [...@default:1] Dial(Local/1...@default-c2a9;2, SIP/1...@100) in new stack [Feb 10 13:31:25] WARNING[3639]: app_dial.c:1712 dial_exec_full: Skipping dialing interface 'SIP/1...@100' again since it has already been dialed i'm calling from 105 to 100 (100 is registred at another domain, defined in sip.conf that's why there is an 1...@100) The device at SIP/100 sent a redirect (forward) message back to Asterisk suggesting that the call be sent to extension '100'. Asterisk refuses to call that device again because it's already been called in that particular instance of Dial and doing so would just result in an infinite loop. You need to figure out why the device at SIP/100 told Asterisk to forward the call when you were expecting it to just accept it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- Groet // Kind regards, Peter den Hartog -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
On Wed, 2010-02-10 at 14:14 +, Brian wrote: On Wed, 2010-02-10 at 08:54 -0500, Ken Leland III wrote: Brian, It could be that the ownership/permissions on the directory are not correct. Are you running asterisk as asterisk:asterisk or root:root? Here is an article that lists the directories and what the ownership/permissions on each one should be: http://www.voip-info.org/wiki/view/Asterisk+non-root Thanks for that, but no - it runs as root, and root can create/access /var/run without any issues. I'm boggled... Answering my own question the solution lies in /etc/asterisk/asterisk.conf: Take the (!) out of /etc/asterisk/asterisk.conf Whilst logic would tell me to replace: astrundir = /var/run to astrundir = /var/run/asterisk ...this did not work. Each time the server is rebooted Asterisk duly deletes the manually created /var/run/asterisk directory - quite why it does this I just don't know - perhaps it is a bug? Leaving: astrundir = /var/run as it is and removing (!) from /etc/asterisk/asterisk.conf does the trick. On rebooting /var/run gets the required .ctl/.pid files and all warnings are banished: srwxr-xr-x 1 root root 0 2010-02-10 14:50 asterisk.ctl -rw-r--r-- 1 root root 5 2010-02-10 14:50 asterisk.pid into /var/run and all warnings vanish of missing .pid and .ctl files are banished. Hopefully this will help someone else. I've seen lots of solutions saying 'create /var/run/asterisk' that have not stood the test of a reboot. -- _ -- 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] IP Phone recommendation
Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Thanks very much in advance, Sebastian Sebastian Milioto ITC Cid Campeadro 440 Rio Tercero, Cordoba, Argentina msn: sebamili...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
On Wed, Feb 10, 2010 at 5:23 AM, RESEARCH resea...@businesstz.com wrote:
snip
You are correct. 8 span which process up to 240 calls at pick time
If the system is actually performing fine then I'd just say that there
is something about the Asterisk threads that makes them look runnable
and that
accounts for the high load average. ?Is the IVR an agi or fastagi or
what? -
I have the agi scripts not as ivr but to help populate the required
information into mysql db. Probably here is where the problem lies i
have
to connect and disconnect to mysql each time a call is made or a
specific
menu is selected
Here is the script
*
#!/usr/bin/perl -w
use strict;
use DBI();
use Scalar::Util qw/weaken/;
my $cdr_log_file = /var/log/asterisk/ivr_log;
my $mysql_host = cdr01;
my $mysql_db = ivrcdrdb;
my $mysql_table = tbl_ivrcdr_details;
my $mysql_user = ivruser;
my $mysql_pwd = a09876a;
my $sth;
my $data0= $ARGV[0];
my $data1= $ARGV[1];
my $data2= $ARGV[2];
my $data3= $ARGV[3];
my $data4= $ARGV[4];
my $data5= $ARGV[5];
my $data6= $ARGV[6];
my $data7= $ARGV[7];
# Connect to database
# print Connecting to database...\n\n;
my $dbh =
DBI-connect(DBI:mysql:database=$mysql_db;host=$mysql_host,$mysql_user,
$mysql_pwd,{'RaiseError'
= 1});
my $insert_str = insert into $mysql_table (calldate, language, src,
duration, accountcode, uniqueid, currentmenu, nextmenu) values
(\$data0\, \$data1\, \$data2\, \$data3\, ?\$data4\,
\$data5\,
\$data6\, \$data7\);\n;
? ? ? $sth = $dbh-prepare($insert_str);
? ? ? $sth-execute();
# print \n\nOK.\n;
$sth-finish();
$dbh-disconnect();
# Trying to resolve memory leak should it happen
delete($ARGV[0]);
delete($ARGV[1]);
delete($ARGV[2]);
delete($ARGV[3]);
delete($ARGV[4]);
delete($ARGV[5]);
delete($ARGV[6]);
delete($ARGV[7]);
exit;
*
the code path may have a spinlock logic to it that means that many
threads
are runnable but when scheduled just go back to sleep. ?That would
account for high load average with lots of spare CPU. ?If that's what is
happening then I wouldn't worry much more about it.
Regards,
Steve
Regards
Sam
If I were you, and I am not and never will be, I would move over to
fastagi and offload all that Perl and database stuff off to a
designated server just to handle that stuff.
I have had the EXACT same problem and that is how it was fixed,
fastagi running to a Windows box that had a process developed (written
in C something) by the M$ developers to hit the M$SQL databases.
We were also doing a ton of things with the AMI which we figured out
how to do the same end result without banging on the AMI, such as
using call files rather than AMI to originate a call.
Load avg dropped to one or under if I remember correctly.
Thanks,
Steve Totaro
Thank you Steve for your recommendation. Ofcoz i have separate server that
is hosting the db and i will consider doing fastagi and see it it will help
@Phil. The credintials displayed there are dummy, so don't worry unless you
mean something else
@Steve Edward. Can you share your C agi codes? I presume what you want me to
do is rewrite the script in C and use it as compiled binary
@Tzafrir. How about this
[ivr4 ~]# ps aux | grep D
USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND
root 1975 0.0 0.0 3920 688 pts/4 S+ 13:17 0:00 grep D
root 3413 0.0 0.0 1832 576 ? Ss 2009 80:58
/usr/sbin/mDNSResponder -b -f /etc/services_mDNS
I have killed that process but no changes
@All, looks like the conclusion has been made that this is to do with AGI.
Let me address it and see how it reacts. I shall feedback
Thanks
Sam
Simple experiment, move to fastagi, perl calls are killing you.
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Re: [asterisk-users] Know what would be killer?
On Thu, Feb 4, 2010 at 7:39 PM, Lyle Underwood lyleunderw...@gmail.com wrote: If call recordings were stored in stereo and the callers were evenly distributed along the stereo spectrum. BAM. Cisco has this. It's called telepresence. It costs a LOT of money, and takes a LOT of bandwidth, but you do get spatial distribution with both video and audio. It requires multiple cameras, multiple monitors, multiple microphones, multiple speakers, but it does work. -- _ -- 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] Optimization of call from server 1 to 2 and thenback to 1
Hi Danny, sorry you are correct: Difficult to say since you don't say if you are on 1.2, 1.4 or 1.6 both Asterisk are running version 1.4.21.2 -- _ -- 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] IP Phone recommendation
Don't use Grandstream if you want quality and stability. Also check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Milioto Sent: Wednesday, February 10, 2010 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IP Phone recommendation Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Thanks very much in advance, Sebastian Sebastian Milioto ITC Cid Campeadro 440 Rio Tercero, Cordoba, Argentina msn: sebamili...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still on spandsp/app_fax and T.38
- Kevin P. Fleming kpflem...@digium.com escreveu: Vinícius Fontes wrote: Will do. You guys will have my feedback on monday. If everything goes okay with that change, I'll post a patch on Mantis. No need for the patch; it's already on my radar, and if you confirm that it actually solves an interop problem, I'll commit the update to the various branches it belongs in. I'd still like to hear from Steve Underwood if I misinterpreted the MMR/JBIG transcoding function calls in spandsp that led me to enabling these features in the first place... -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org - Vinícius Fontes vinic...@canall.com.br escreveu: Unfortunely it didn't solve the problem. Here's the session packet capture after editing app_fax.c. http://www.canall.com.br/wireshark_t38_jbig.gz Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 Sorry for the shameless bump, but... any news on this? :) -- _ -- 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] IP Phone recommendation
I was recommended Polycom phones. I tested some. And now, I LOVE them. Look at the Polycom IP321. It's a great phone with provisioning and two lines. Dont know about G729, but I'd be surprised if it didn't support it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder Sent: 10 February 2010 15:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] IP Phone recommendation Don't use Grandstream if you want quality and stability. Also check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Milioto Sent: Wednesday, February 10, 2010 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IP Phone recommendation Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Thanks very much in advance, Sebastian Sebastian Milioto ITC Cid Campeadro 440 Rio Tercero, Cordoba, Argentina msn: sebamili...@hotmail.commailto:sebamili...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
On 10 Feb 2010, at 15:50, Peder wrote: check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. I second that. They're rock solid, good audio quality and easy to provision. S -- _ -- 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] EAGI delay
Hello, I made a post to the forums (http://forums.digium.com/viewtopic.php?f=1t=72901sid=3d5c2717ca5ab7ad676957ae436d4b51) but haven't received any replies, so thought I'd try here. On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been noticing that there's a problem with conferences (using both meetme and app_conference) and the audio sent out to an EAGI script. My setup is essentially a conference which then periodically gets the EAGI application run in it. What I've noticed is that, when the conference is newly-created, there is no latency (or at least, minimal latency). After the conference has been running for a while though, the delay in the audio sent to the EAGI script increases dramatically. After 45 minutes, the EAGI script gets audio that's over 20 seconds out of sync! This was tested with an extremely simple eagi script: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw the audio was then converted with: sox -t raw -r 8000 -w -s -c 1 - output.wav The bug doesn't seem to be in the eagi end of things though, because if I let the conference run for a while with no eagi in the conference, and then add it in after a while, I see the delay. It looks like this is a bug in asterisk. Is there any known workaround? Any chance it might be fixed in asterisk 1.6? (I plan on testing this out, but so far I haven't been able to get app_conference working in it, and meetme won't work since my test server is in a xen domU, so I have no timing source) Also, I noticed some lines in the log which may be related: [Jan 21 15:12:46] WARNING[8574] conference.c: processed frame frequency variation, name = ConferenceA_test, tf_count = 50, tf_diff = 950, tf_frequency = 19. There are several lines like this, with tf_diff varying between 912-1083, and tf_frequency from 18.24-21.66. Does anyone know what this might mean? Thanks for any help you might be able to offer! -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] IP Phone recommendation
Polycom 331's are also in the same price range, and offer good features as well. All my polycoms are provisions with option 66 on dhcp, and an ftp site with cfg files that are build from a mysql database from sip users table. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder Sent: Wednesday, February 10, 2010 10:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] IP Phone recommendation Don't use Grandstream if you want quality and stability. Also check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Milioto Sent: Wednesday, February 10, 2010 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IP Phone recommendation Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Thanks very much in advance, Sebastian Sebastian Milioto ITC Cid Campeadro 440 Rio Tercero, Cordoba, Argentina msn: sebamili...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 I'd like to jump in here as well, with the Aastra 57i. It is easy to configure with asterisk, provision and is not that badly priced either. - - Tommy William Stillwell (Lists) skrev: Polycom 331’s are also in the same price range, and offer good features as well. All my polycoms are provisions with option 66 on dhcp, and an ftp site with cfg files that are build from a mysql database from sip users table. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Peder *Sent:* Wednesday, February 10, 2010 10:51 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] IP Phone recommendation Don’t use Grandstream if you want quality and stability. Also check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sebastian Milioto *Sent:* Wednesday, February 10, 2010 9:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] IP Phone recommendation Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Thanks very much in advance, -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkty3gsACgkQ573V05EH/pZOUwCfdwbZD1Bs+PG1iD4WBWwaP3KL 0+wAn3pysUcluzjjcW43hqTa1JSlEwbf =0pyC -END PGP SIGNATURE- -- _ -- 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 sudden restart - 1.4.18.1
Hi, Asterisk got stopped this morning after 20 minutes and phones went to 'No Service' and then got started automatically after 20 min, as I could see in the full log that asterisk got started at so and so time: [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Dynamic Loader Starting: But I am trying to find why did it stopped (and there was no record of asterisk stopped?) and then get restarted.In the log I could also see : [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:08] VERBOSE[7004] logger.c: -- Remote UNIX connection [Feb 10 08:02:08] VERBOSE[28272] logger.c: -- Remote UNIX connection disconnected During this period (from 8:02 till 8:29) all the phones went to 'No Service'I checked all the logs and could not find any reason why it was down or any log that shows asterisk was down at that point..any ideas are appreciated... Asterisk version: 1.4.18.1 Thanks so much Regards Sandesh -- _ -- 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] IP Phone recommendation
On 10.2.2010 18:06, Steve Howes wrote: On 10 Feb 2010, at 15:50, Peder wrote: check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. I second that. They're rock solid, good audio quality and easy to provision. S SPA504G - 1 more vote for it. It is worth having 4 lines even if you need 1 initially. SPA504G supports G722 and sound is awesome even if you do not not use teh HD sound. If you do not care that mcuh about HD sound and do not need PoE SPA941 is a excellent choice - you get really a lot for the price Peter -- _ -- 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] Muted calls occasionally dropping after 30 seconds
Ishfaq- I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. Are you using a codec (such as G729) on the outgoing leg of that line? If so you might check for VAD/DTX enabled and see if that makes any difference. -Jeff The rtptimeout and rtpholdtimeout are both set to 0 on a global level and for the sip extension the sip table row has NULL in both columns. I've tried playing with those 2 values, both on a global and sip extension level but regardless to what they are set to, if the call gets disconnected it is always 30 seconds after the mute button is pressed. But like I said before, this does not happen every time the mute button is pressed. I managed to recreate the phenomenon one one of our test servers so I could be certain that there was nothing else going on at the time. The call path when recreating this on our test platform was My Mobile - number/SIP provider - out asterisk server - SIP extension Has anyone else ever experienced anything like this? It's really got me rather frustrated! Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] 1.6.2 : global vars not read/set after #include w/ globals
I found out that the [globals] section in extensions.conf is ignored if an #include 'd file has a [globals] section. Is this intended? In this particular case, the #include 'd file has a number of contexts for googlevoice. I'd put various googlevoice variables in there to use in all those contexts. Once I did that all of the global variables set in extensions.conf were ignored. sean -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
Brian wrote: Each time the server is rebooted Asterisk duly deletes the manually created /var/run/asterisk directory - quite why it does this I just don't know - perhaps it is a bug? Your assumption is incorrect. Some Linux distributions will empty /var/run/ on boot, just as they do with /tmp/. I do believe you're right, however, in suggesting that there is a bug in Asterisk. It appears that Asterisk creates /var/run/asterisk/ during install and assumes that it will always exist. Some of the sample init scripts (Debian) create that directory before starting Asterisk. This should be done in all of them (or in Asterisk itself, maybe?). Please report an issue on http://issues.asterisk.org/ -- _ -- 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] IP Phone recommendation
I see... problem with spa941 is it dont have LAN port (I'm thinking NAT the customer PC) Sebastian On Wed, Feb 10, 2010 at 2:10 PM, Peter peterp...@aboutsupport.com wrote: On 10.2.2010 18:06, Steve Howes wrote: On 10 Feb 2010, at 15:50, Peder wrote: check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. I second that. They're rock solid, good audio quality and easy to provision. S SPA504G - 1 more vote for it. It is worth having 4 lines even if you need 1 initially. SPA504G supports G722 and sound is awesome even if you do not not use teh HD sound. If you do not care that mcuh about HD sound and do not need PoE SPA941 is a excellent choice - you get really a lot for the price Peter -- _ -- 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 -- _ -- 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] IP Phone recommendation
SPA504G - 1 more vote for it. It is worth having 4 lines even if you need 1 initially. SPA504G supports G722 and sound is awesome even if you do not not use teh HD sound. If you do not care that mcuh about HD sound and do not need PoE SPA941 is a excellent choice - you get really a lot for the price Peter Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced SIP compatibility with asterisk more completely with the SPA504G's than they have the 7940 series? Lack of features on the 7940's is frustrating, and makes me hesitant to try other Cisco phones, even if the SPA504G is newer. --Brent -- _ -- 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] IP Phone recommendation
On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga li...@torrenga.com wrote: Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced SIP compatibility with asterisk more completely with the SPA504G's than they have the 7940 series? Lack of features on the 7940's is frustrating, and makes me hesitant to try other Cisco phones, even if the SPA504G is newer. SIP support in newer generations of the 79XX series is much better. I believe that their goal is to have 100% feature parity between the SIP and the SCCP images, they are probably 90% now. Whether Asterisk supports all of those features is another matter though. -- Jeff Ollie -- _ -- 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] How to run a remote PHP script and still have access to audio stream?
Hello Ben, thanks for your message! I'm implementing a framework to integrate Asterisk and Drupal (a powerful tool for the creation of social- and media-rich websites). Since the voip and the web components of the system are likely to run on separate servers owned by different organizations, I don't think I could rely on a shared filesystem solution. Currently, I save Asterisk audio files on the Asterisk server, convert them to MP3, upload them to the Drupal server, and display them on the website. This process takes time and ends up duplicating a lot of content. That's why I was exploring ways of doing all the audio processing on the Drupal server with FastAGI and EAGI. Is there any easy way for Asterisk to play audio files located in remote servers? Another alternative would be to get Drupal to play audio files directly from the Asterisk server. Would you have any suggestions for that? Thanks once again for all your support! Leo Ben Dinnerville wrote: There is the EAGI protocol that will allow this but the easiest way I find to do this sort of thing is to have a shared file system between the app / web server and the asterisk server(s). We run a clustered setup with 12 asterisk systems and a clustered jboss environment with a NFS mount shared between all the systems. For applications such as call recording asterisk does the monitor into the NSF mounted share / directory which is also visible on the jboss servers (mainly for permission checks / security etc) and the web server (for download etc). As long as you have a scheme that ensures you do not have duplicate file names (which can be controlled by a central database and via your php script) then you will not have any issues. There are a number of other file system alternatives out there that will achieve the same thing but NFS seems to be proven and stable and we have not any issues with it to date. Your php script can then be a simple AGI / FastAGI that simply executes a Playback(path/to/nfs/directory/file) - you can also incorporate things like checking if the file exists in your PHP script and implementing access restrictions etc. We also share our sounds directory between systems this way so that all our sounds only have to reside in one place but are visible across all the systems. Cheers, Ben Leo Burd wrote: Hello there, I'm trying to figure out how to run a PHP script on a remote machine and still have access to the audio stream associated with the call. Ideally, I'd love to play/record audio files directly from/to the remote server without having to copy them back and forth to the Asterisk server. What is the best way to do this? Is it possible to combine EAGI with FastAGI in PHP? Thanks in advance, Leo -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
On Wed, 2010-02-10 at 11:24 -0600, Jason Parker wrote: Brian wrote: Each time the server is rebooted Asterisk duly deletes the manually created /var/run/asterisk directory - quite why it does this I just don't know - perhaps it is a bug? Your assumption is incorrect. Some Linux distributions will empty /var/run/ on boot, just as they do with /tmp/. Thanks Jason - that had never dawned on me, but I've just tested it and indeed it does. I do believe you're right, however, in suggesting that there is a bug in Asterisk. It appears that Asterisk creates /var/run/asterisk/ during install and assumes that it will always exist. Agreed - that would make sense that by default it thinks the directory is there. The workaround / fix is to take out the (!) from /etc/asterisk/asterisk.conf and allowing the default setting of: astrundir = /var/run to come into play. It then puts the .pid and .ctl in the root of /var/run Some of the sample init scripts (Debian) create that directory before starting Asterisk. This should be done in all of them (or in Asterisk itself, maybe?). The one I had didn't - but I could have added it. I just wanted to be sure I was doing the right thing. Please report an issue on http://issues.asterisk.org/ Done - but I'm a bit embarrassed as it seems so trivial. Thank you for your help. -- _ -- 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] Security Logging
On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese l...@lcrcomputer.net wrote: Here's a start for you, just run from cron once a day: Lyle So basically, nothing built into asterisk that already provides security logging mechanisms? Maybe I'm using the wrong term; In Windows, I think it would be called Security Auditing, successful / unsuccessful login attempts that get recorded in the Windows Event Viewer in the security log. These login attempts (whether successful or not) are recorded, and you get the IP address of the workstation attempting the login, the username used, and whether or not it was successful. A log dedicated just to security auditing (or a new option in /etc/logger.conf that adds this functionality (say, messages = notice,warning,error,verbose,security) seems like it would be a nice addition to asterisk. I've already got tools that can monitor log files and create bans based on failed login attempts...but I don't always seem to see login failures in the asterisk messages log. I recall from Astricon 2009, Russel and Kevin (I think) commenting on security features in asterisk and not sure how much to include (i.e automatically banning people based on failed login attempts being a process asterisk controls or just simply logs so that another tool can do the banning, etc). I just don't remember if there was any followup to those discussions. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?
On Wed, 2010-02-10 at 13:47 -0500, Leo Burd wrote: Is there any easy way for Asterisk to play audio files located in remote servers? If you can mount it, Asterisk will happily read from it. Perhaps you can run a an ssh/ftp/smb/nfs server deamon on the webserver and mount it on the filesystem on the Asterisk box? I'd probably want to store them locally and have a script check the remote storage is online and then move them. A simple FTP script may be the easiest way to achieve this. Another alternative would be to get Drupal to play audio files directly from the Asterisk server. Would you have any suggestions for that? Host them both on the same machine :-) Der dum chishh. I'll get my coat. -- _ -- 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] Nat Issue - is this Draytek || Asterisk?
On Wed, Feb 10, 2010 at 5:53 AM, Brian brel.astersik100...@copperproductions.co.uk wrote: I'm struggling to work out how can I debug this effectively and would appreciate some guidance here. Try enabling sip debug on the internal peers (sip set debug peer from the cli) before you bring the second peer up, and go from there? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?
Hello David, Thanks so much for your message! Please check my comments inline below... David Backeberg wrote: On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd l...@media.mit.edu wrote: Hello there, I'm trying to figure out how to run a PHP script on a remote machine and still have access to the audio stream associated with the call. Ideally, I'd love to play/record audio files directly from/to the remote server without having to copy them back and forth to the Asterisk server. What is the best way to do this? Is it possible to combine EAGI with FastAGI in PHP? You don't specify how often / what proportion of the recordings need to be on a remote machine versus on the asterisk server. So you have two main things going on: 1) recordings, with a side order of distributing those to another machine 2) remote shell scripting First, the recordings can be done directly on a channel where the call is taking place. If this is one call, that's not so bad, but there get to be I/O contention issues when you try to record 'a lot' of calls simultaneously. Some people endorse working around that by writing recordings to a ramdisk, and then occasionally flushing those off to a real hard disk. What would be the asterisk way of recording part of the call from a remote server? I'm not sure I can do that (the remote connection) with EAGI, can I? You may prefer an alternate approach, which is that taken by commercial recording solutions. Oreka (which can be grabbed from sourceforge), and pretty much every commercial voip recording solution I've investigated, works by having you use libpcap (used in Ethereal/Wireshark) to watch ethernet device(s) where voip calls are taking place, grab the SIP headers that set up the RTP stream, and then write those recordings to disk on a dedicated recordings server. This requires explicit ethernet support by doing things like port mirroring, or using an old-school hub, etc. This has an advantage for you of providing a way to do recording directly on a machine that is NOT the asterisk server. No copying required as the recording is already where you want it. Second, the remote shell isn't so hard. ssh with keys, problem solved. You can do that directly from the asterisk dialplan using the System() command. This let's you tie the remote shell directly to a given call, where you can tune arguments accordingly. Do you know of any examples that use ssh from inside Asterisk calls? How much control do the ssh processes have over the call, if any? Is that comparable to Fast_AGI? Or EAGI? Of course, you can also do #1 with scripting and remote shell, or rsync with keys. If you don't need 'a lot' of simultaneous channels recorded, this may be more straightforward. You only have to learn asterisk, rather than asterisk and Oreka. I'm intrigued about the remote shell idea... please let me know if you have additional information about it, ok? Thanks once again, Leo -- _ -- 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] IP Phone recommendation
On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga li...@torrenga.com wrote: Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced SIP compatibility with asterisk more completely with the SPA504G's than they have the 7940 series? Lack of features on the 7940's is frustrating, and makes me hesitant to try other Cisco phones, even if the SPA504G is newer. --Brent I talked to the Cisco SPA guy at the 2009 Astricon convention about this - according to him, the SPA line is a completely different team than the 79xx series team. Thus, the SPA (and especially the new SPA5xx series) had better support for SIP out of the box, because that's what they were originally designed for (whereas the 79xx series was built for SCCP and CUCM). There was even a document (https://www.myciscocommunity.com/docs/DOC-10647) that listed how to make the SPA5xx series phones work with asterisk. The phones were really nice, but I haven't seen one outside of a demonstration environment. As far the OP's question, I've successfully used Polycom, Cisco, and Aastra phones, all in the price range listed, that have been very nice and support all the options you've requested. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems with 1.6
In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating my EAGI script, etc) the log fills up with error messages and the channel disconnects immediately. What could be wrong? My configuration is mostly duplicated from the old 1.4 config - has something perhaps changed there that's causing the problem? I've looked at various upgrade instructions and haven't seen anything, but it's easy to miss details in the reams of info out there! In any case, the log looks like this: == Manager 'veco' logged on from 127.0.0.1 [Feb 10 14:14:36] ERROR[15569]: res_timing_timerfd.c:105 timerfd_timer_open: Failed to create timerfd timer: Function not implemented [Feb 10 14:14:36] ERROR[15569]: res_timing_timerfd.c:105 timerfd_timer_open: Failed to create timerfd timer: Function not implemented -- Executing [confere...@veco:1] Answer(Local/confere...@veco-044d;2, ) in new stack [Feb 10 14:14:37] WARNING[15571]: channel.c:1065 __ast_queue_frame: Unable to write to alert pipe on Local/confere...@veco-044d;1 (qlen = 0): Broken pipe! [Feb 10 14:14:37] WARNING[15571]: channel.c:1065 __ast_queue_frame: Unable to write to alert pipe on Local/confere...@veco-044d;1 (qlen = 1): Broken pipe! -- Executing [confere...@veco:2] NoOp(Local/confere...@veco-044d;2, Trying to start conference ConferenceA_test) in new stack -- Executing [confere...@veco:3] Konference(Local/confere...@veco-044d;2, ConferenceA_test) in new stack [Feb 10 14:14:38] WARNING[15571]: channel.c:1065 __ast_queue_frame: Unable to write to alert pipe on Local/confere...@veco-044d;1 (qlen = 2): Broken pipe! [Feb 10 14:14:38] WARNING[15571]: channel.c:1065 __ast_queue_frame: Unable to write to alert pipe on Local/confere...@veco-044d;1 (qlen = 3): Broken pipe! [repeated many, many times] [Feb 10 14:14:40] WARNING[15571]: channel.c:1045 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/confere...@veco-044d;1 [Feb 10 14:14:40] WARNING[15571]: channel.c:1045 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/confere...@veco-044d;1 [also repeated many, many times] -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
The other way on Debian/Ubuntu is just to test the existence of the dir and create it if needed If you add this to the /etc/init.d/asterisk near the start you should be fine if ! [ -d /var/run/asterisk ] ; then mkdir /var/run/asterisk chown $AST_USER.$AST_GROUP /var/run/asterisk exit 0 fi Set the ownership as required Cheers Duncan On 11/02/2010, at 7:50 AM, Brian wrote: On Wed, 2010-02-10 at 11:24 -0600, Jason Parker wrote: Brian wrote: Each time the server is rebooted Asterisk duly deletes the manually created /var/run/asterisk directory - quite why it does this I just don't know - perhaps it is a bug? Your assumption is incorrect. Some Linux distributions will empty /var/run/ on boot, just as they do with /tmp/. Thanks Jason - that had never dawned on me, but I've just tested it and indeed it does. I do believe you're right, however, in suggesting that there is a bug in Asterisk. It appears that Asterisk creates /var/run/asterisk/ during install and assumes that it will always exist. Agreed - that would make sense that by default it thinks the directory is there. The workaround / fix is to take out the (!) from /etc/asterisk/asterisk.conf and allowing the default setting of: astrundir = /var/run to come into play. It then puts the .pid and .ctl in the root of /var/run Some of the sample init scripts (Debian) create that directory before starting Asterisk. This should be done in all of them (or in Asterisk itself, maybe?). The one I had didn't - but I could have added it. I just wanted to be sure I was doing the right thing. Please report an issue on http://issues.asterisk.org/ Done - but I'm a bit embarrassed as it seems so trivial. Thank you for your help. -- _ -- 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 -- _ -- 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] billing based on local access number
At 4:02 AM on 10 Feb 2010, umesh maharjan wrote:
Hi all,
I am configuring asterisk as a prepaid calling card. I am getting
different local rate from my ISDN provider e.g 0.002 for landline
and 0.13 for mobile etc. In this case I thing I have to say my
asterisk/a2billing to bill based on local access number. so How can I
retrieve called number (eg. 03-6832-1040 and 0120-272-060 is our
ISDN PRI access number) to my asterisk server so i can trigger
different rates.
The number the caller called to get to you should be passed to Asterisk
as the inbound extension. So, in your incoming context, you can
provide different extensions for the different incoming numbers. Or
you can catch everything with the _X. pattern and use the ${EXTEN}
variable to check the number in your dialplan.
One thing to note is that it doesn't always pass the whole number. I
have two PRIs from different providers; one of them passes all 10
digits, but the other one only passes the last 4, and for some reason
with one of our numbers that ends in 9977 the PRI passes 2977. You
can either ask your provider what they pass, or you can just make test
calls and log the value of the ${EXTEN} variable with Verbose() calls,
something like this:
[incoming]
exten = _X.,1,Verbose(Incoming call to ${EXTEN});
exten = _X.,n,Playback(welcome);
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
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Re: [asterisk-users] IP Phone recommendation
On Wed, 10 Feb 2010, Sebastian Milioto wrote: Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Since you've had one negative for the Grandstream, I'll balance it and give them a positive... Now, it's true to say that Grandstream phones haven't been without their probems in the past, but the current generation are really nice. The GXP2000 has been about for a long time too and I've not had any issues with them in the past 18 months or so. They've undergone a few hardware revisions too. I statically provision all my phones and use a perl utility called gsutil. They can be provisioned from a tftp server though, but I've never done this. They sound fine, and have plenty of features that are easy to use - call transfer - big backlit display (now variable contrast and brigtness) 7 easy to use speed dial/BLF buttons, etc. If not using PoE I'd suggest getting a few extra PSUs though - that's one area I have had a few issues with - but maybe it's just been the UK ones. Gordon -- _ -- 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] IP Phone recommendation
SPA504G is LINKSYS with newer look and HD :-) Expect all you had in Linksys SPA9XX + more. I personaly have both phones - differences are not a lot :) Peter On 10.2.2010 20:31, Jeffrey Ollie wrote: On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga li...@torrenga.com wrote: Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced SIP compatibility with asterisk more completely with the SPA504G's than they have the 7940 series? Lack of features on the 7940's is frustrating, and makes me hesitant to try other Cisco phones, even if the SPA504G is newer. SIP support in newer generations of the 79XX series is much better. I believe that their goal is to have 100% feature parity between the SIP and the SCCP images, they are probably 90% now. Whether Asterisk supports all of those features is another matter though. -- _ -- 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] IP Phone recommendation
- Gordon Henderson gordon+aster...@drogon.net wrote: If not using PoE I'd suggest getting a few extra PSUs though - that's one area I have had a few issues with - but maybe it's just been the UK ones. Gordon The same can be said for the US versions. My experience has been it's not a case of 'if' the PSU will fail, but 'when'. In a past (less intelligent) life, I deployed a fair number of the GXP2020s and GXP2000s. There are not very many of them left that haven't completely died(the phone itself), and of those left, they've all had power supplies replaced. I cannot speak for the quality of the later devices from Grandstream. After being burned, it's a bit hard to look at them again when there are so many other quality devices available (think Polycom, Aastra, etc). --Tim -- _ -- 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] IP Phone recommendation
Hi! Lack of features on the 7940's is frustrating, and makes me hesitant to try other Cisco phones, even if the SPA504G is newer. Here are two quotes that make me stay away from Cisco/Linksys: Firmware can be downloaded from the Cisco Support Center (registered partners only - password required) [...] Here is a 96-second screencast showing navigation to the firmware: http://screencast.com/t/CUnxfoAX; A 1.5 min screencast to explain how to download firmware?! ;-) Philipp -- _ -- 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] IP Phone recommendation
On Wed, 10 Feb 2010, Tim Nelson wrote: - Gordon Henderson gordon+aster...@drogon.net wrote: If not using PoE I'd suggest getting a few extra PSUs though - that's one area I have had a few issues with - but maybe it's just been the UK ones. Gordon The same can be said for the US versions. My experience has been it's not a case of 'if' the PSU will fail, but 'when'. In a past (less intelligent) life, I deployed a fair number of the GXP2020s and GXP2000s. There are not very many of them left that haven't completely died(the phone itself), and of those left, they've all had power supplies replaced. I cannot speak for the quality of the later devices from Grandstream. After being burned, it's a bit hard to look at them again when there are so many other quality devices available (think Polycom, Aastra, etc). --Tim I haven't used any standard Grandstream IP phones, but I am *trying* to stabalize the new video phones they have come up with. I have several GXV3000 and GXV3140s. I got through central provisioning using their java based tool and for the most part these phones work, but have very odd bugs. If left to itself for more than a few days the 3140 simply stops answering calls. The 3000 has very odd DTMF issues - like doubling every digit pressed. This is all fine and I know they are new products, but what is frustrating is Grandstream's lack of support. The forums are next to useless, and the firmware releases are always coming very soon. Then there are my horrid experiences with their FXO gateways. Echo, bad audio in general, needing a reboot every few days, etc. Again, support is non existant. So regardless of the quality of the latest phones, the company itself leaves a lot to be desired IMO. j -- _ -- 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] 1.6.2 : global vars not read/set after #include w/ globals
sean darcy wrote: I found out that the [globals] section in extensions.conf is ignored if an #include 'd file has a [globals] section. Is this intended? In this particular case, the #include 'd file has a number of contexts for googlevoice. I'd put various googlevoice variables in there to use in all those contexts. Once I did that all of the global variables set in extensions.conf were ignored. Context names cannot be duplicated, unless you suffix them with (+) to allow them to be added together. It does not matter whether it is the 'global' context or any other context. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] IP Phone recommendation
On Wed, Feb 10, 2010 at 3:21 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Here are two quotes that make me stay away from Cisco/Linksys: Firmware can be downloaded from the Cisco Support Center (registered partners only - password required) [...] Here is a 96-second screencast showing navigation to the firmware: http://screencast.com/t/CUnxfoAX; A 1.5 min screencast to explain how to download firmware?! ;-) The SPA firmware only requires free registration - you do not need to have a SmartNET contract to get the firmware (according to their site and the conversation I had with the Cisco SPA rep at Astricon). -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
I would love to hear some inputs on Aastra and Snom IP phones. On Wed, Feb 10, 2010 at 4:36 PM, Jeff LaCoursiere j...@jeff.net wrote: On Wed, 10 Feb 2010, Tim Nelson wrote: - Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: If not using PoE I'd suggest getting a few extra PSUs though - that's one area I have had a few issues with - but maybe it's just been the UK ones. Gordon The same can be said for the US versions. My experience has been it's not a case of 'if' the PSU will fail, but 'when'. In a past (less intelligent) life, I deployed a fair number of the GXP2020s and GXP2000s. There are not very many of them left that haven't completely died(the phone itself), and of those left, they've all had power supplies replaced. I cannot speak for the quality of the later devices from Grandstream. After being burned, it's a bit hard to look at them again when there are so many other quality devices available (think Polycom, Aastra, etc). --Tim I haven't used any standard Grandstream IP phones, but I am *trying* to stabalize the new video phones they have come up with. I have several GXV3000 and GXV3140s. I got through central provisioning using their java based tool and for the most part these phones work, but have very odd bugs. If left to itself for more than a few days the 3140 simply stops answering calls. The 3000 has very odd DTMF issues - like doubling every digit pressed. This is all fine and I know they are new products, but what is frustrating is Grandstream's lack of support. The forums are next to useless, and the firmware releases are always coming very soon. Then there are my horrid experiences with their FXO gateways. Echo, bad audio in general, needing a reboot every few days, etc. Again, support is non existant. So regardless of the quality of the latest phones, the company itself leaves a lot to be desired IMO. j -- _ -- 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 -- Pascal B. http://www.kameleonlabs.com/ Samuel Goldwynhttp://www.brainyquote.com/quotes/authors/s/samuel_goldwyn.html - I'm willing to admit that I may not always be right, but I am never wrong. -- _ -- 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] IP Phone recommendation
I would love to hear some inputs on Aastra and Snom IP phones. I'm using Aastra 57i phones and like them. They can provisioned easily (without ANY entries from a local network). The support BLF and I'm also using the XML capability. -- _ -- 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] How to avoid AGI script is canceled if caller HangUp
Hi, is there any way to avoid cancel the AGI script if caller is hanging up. That gives me sometimes data mismatch and it is deffcault to clean up in the h extension. I would like that the PHP script called by AGI will run to end.. Some thing can happend with an Macro if caller hang up exactly when call is answered. An Macro called byi the DIAL command will be stoped and data mismatch can occur.. best regards Thomas -- _ -- 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] How to avoid AGI script is canceled if callerHangUp
According to the CLI doc, you can do it this way - exten = 100,1,Set(AGISIGHUP=no) - exten = 100,n,AGI(youragi.agi) YMMV -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Winter Sent: Wednesday, February 10, 2010 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to avoid AGI script is canceled if callerHangUp Hi, is there any way to avoid cancel the AGI script if caller is hanging up. That gives me sometimes data mismatch and it is deffcault to clean up in the h extension. I would like that the PHP script called by AGI will run to end.. Some thing can happend with an Macro if caller hang up exactly when call is answered. An Macro called byi the DIAL command will be stoped and data mismatch can occur.. best regards Thomas -- _ -- 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 -- _ -- 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] How to avoid AGI script is canceled if callerHangUp
Un-top-posting... [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Winter is there any way to avoid cancel the AGI script if caller is hanging up. That gives me sometimes data mismatch and it is deffcault to clean up in the h extension. I would like that the PHP script called by AGI will run to end.. On Wed, 10 Feb 2010, Danny Nicholas wrote: According to the CLI doc, you can do it this way - exten = 100,1,Set(AGISIGHUP=no) - exten = 100,n,AGI(youragi.agi) Who knew? Hey TP, I learned something new today! Another approach, is to establish a signal handler -- so you can handle the signal :) I write my AGIs in C (because you can execute xxx AGIs written in C in the time it takes to load PHP and parse your script) so it looks like this: // trap SIGHUP -- caller hung up signal(SIGHUP, (void (*)(int))(int)hangup); When the caller hangs up, Asterisk delivers a SIGHUP to the process created by the agi() application. Execution of your AGI will then continue with your signal handler where you can clean up temporary files, roll back database cruft, etc. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to avoid AGI script is canceled if callerHangUp
On Wednesday 10 February 2010 17:13:09 Steve Edwards wrote: Un-top-posting... [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Winter is there any way to avoid cancel the AGI script if caller is hanging up. That gives me sometimes data mismatch and it is deffcault to clean up in the h extension. I would like that the PHP script called by AGI will run to end.. On Wed, 10 Feb 2010, Danny Nicholas wrote: According to the CLI doc, you can do it this way - exten = 100,1,Set(AGISIGHUP=no) - exten = 100,n,AGI(youragi.agi) Who knew? Hey TP, I learned something new today! Another approach, is to establish a signal handler -- so you can handle the signal :) I write my AGIs in C (because you can execute xxx AGIs written in C in the time it takes to load PHP and parse your script) so it looks like this: // trap SIGHUP -- caller hung up signal(SIGHUP, (void (*)(int))(int)hangup); When the caller hangs up, Asterisk delivers a SIGHUP to the process created by the agi() application. Execution of your AGI will then continue with your signal handler where you can clean up temporary files, roll back database cruft, etc. One thing that you cannot do in 1.4 is continue to interact with the AGI interface, however. You can handle the signal, but once it is sent, you can no longer interact with Asterisk. This deficiency is fixed in the 1.6 series. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
At 02:18 PM 2/10/2010, you wrote: I would love to hear some inputs on Aastra and Snom IP phones. I've have 3 480i-CT Aastra phones in our house for 3 or 4 years now with no complaints. Took a year for the firmware to get where it is and there were some things I'd like changed, but I can't remember what they are any more. Other than loosing 1 cordless handset they've been rock solid. Ira -- _ -- 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] wellgate 3804A with frying
Dear Colleagues, I installed a Wellgate 3804A and overnight lines on all this with frying, putting other lines Wellgate 3804A is well, so I guess it's a problem the first team which is already out of warranty, anyone know how can I fix this? or where to send it in or capital Buenos Aires to fix it? Thanks Mart _ Todo lo que querés saber sobre la TV y sus protagonistas en MSN http://msn.novebox.com/-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
On Thu, Feb 11, 2010 at 08:45:05AM +1300, Duncan Turnbull wrote: The other way on Debian/Ubuntu is just to test the existence of the dir and create it if needed If you add this to the /etc/init.d/asterisk near the start you should be fine if ! [ -d /var/run/asterisk ] ; then mkdir /var/run/asterisk chown $AST_USER.$AST_GROUP /var/run/asterisk Please use ':' as a separator for chown. chown $user: file # (empty group) chowns the file to the default group of the uiser, which means that it's safe to leave AST_GROUP set to an empty value. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending Progress during dialing
The PBX that I'm connecting to Asterisk has a timeout on calls on its PRI and QSIG lines. But that's smaller than the time it can take some SIP trunk providers to complete the calls, so I get hangups. I verified that sending Progress every few seconds will work around the problem. So I'd like to see Dial do that. I don't see any mechanism though it appears that it's fairly easy to write one. Is there something I'm missing: can this be done without any code changes? Is there any reason why the obvious code change wouldn't work? Is there any harm from a specification point of view in sending multiple Progress messages? -- _ -- 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] 1.6.2 : global vars not read/set after #include w/ globals
Kevin P. Fleming wrote: sean darcy wrote: I found out that the [globals] section in extensions.conf is ignored if an #include 'd file has a [globals] section. Is this intended? In this particular case, the #include 'd file has a number of contexts for googlevoice. I'd put various googlevoice variables in there to use in all those contexts. Once I did that all of the global variables set in extensions.conf were ignored. Context names cannot be duplicated, unless you suffix them with (+) to allow them to be added together. It does not matter whether it is the 'global' context or any other context. Well Dialplan reloaded. == Parsing '/etc/asterisk/extensions.conf': == Found .. == Parsing '/etc/asterisk/exts/gvoice.exten.conf': == Found cat exts/gvoice.exten.conf [+globals] test-global = need-a-plus-sign . but no test-global in dialplan show globals :( sean -- _ -- 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] 1.6.2.1: DTMF trouble with PSTN
sean darcy wrote: Tzafrir Cohen wrote: On Fri, Feb 05, 2010 at 01:55:03PM -0500, sean darcy wrote: sean darcy wrote: Using 1.6.2.1 with a TDM400, attached to internal analog phones and PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for something stupid. The call itself works, but the DTMF tones fail. -- Starting simple switch on 'DAHDI/1-1' -- Executing [6258...@internal:1] Answer(DAHDI/1-1, ) in new stack -- Executing [6258...@internal:2] Dial(DAHDI/1-1, DAHDI/4/ww2156258013) in new stack -- Called 4/ww2156258013 -- DAHDI/4-1 answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/4-1 -- Hungup 'DAHDI/4-1' Any suggestions? sean This is DAHDI Tools Version - 2.2.1 Do DTMF tones work for others over dahdi? I'd file a bug, but I'd like to make sure it's not just my mistake. Do DTMFs work on a simple call to Asterisk? A simple IVR or VoiceMail. Can you recerd the audio before it gets to Asterisk? use: dahdi_monitor 1 -r rec.raw; sox -r 8000 -c 1 -s -w rec.wav Can you hear the DTMFs in rec.wav? Another thing to try: press a key for a few seconds. Do you hear it continously? I didn't understand how to actually do what you asked :( , but here's what I did do: I set up the dial plan to use sip for a local number, rather than just dial out on PSTN over dahdi. That worked. So that must mean the DTMF tones get to asterisk over the internal dahdi channel, but for some reason are not sent out over the outgoing dahdi channel, right? I do not hear a continuous tone if I press a key for a few seconds. sean Any thoughts, or should I file a bug? If it is a bug, it's dahdi, right? sean -- _ -- 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] Security Logging
Warren Selby wrote: On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: Here's a start for you, just run from cron once a day: Lyle So basically, nothing built into asterisk that already provides security logging mechanisms? Maybe I'm using the wrong term; In Windows, I think it would be called Security Auditing, successful / unsuccessful login attempts that get recorded in the Windows Event Viewer in the security log. These login attempts (whether successful or not) are recorded, and you get the IP address of the workstation attempting the login, the username used, and whether or not it was successful. A log dedicated just to security auditing (or a new option in /etc/logger.conf that adds this functionality (say, messages = notice,warning,error,verbose,security) seems like it would be a nice addition to asterisk. I've already got tools that can monitor log files and create bans based on failed login attempts...but I don't always seem to see login failures in the asterisk messages log. I recall from Astricon 2009, Russel and Kevin (I think) commenting on security features in asterisk and not sure how much to include (i.e automatically banning people based on failed login attempts being a process asterisk controls or just simply logs so that another tool can do the banning, etc). I just don't remember if there was any followup to those discussions. -- Thanks, --Warren Selby http://www.selbytech.com I think that is the problem. Nobody can agree on how it should be implemented. So just log the events and the user/admin find and use a log analyzer or build your own tools for those that want/need such. Lyle -- _ -- 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] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a outgoing call was made, we got the following error message: app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) We suspect it was the ISDN line problem and so we waited a whole day for the engineer to arrive. He plugged an ISDN phone into the line and found it was working because he could call out. We are perplexed and thought about replacing the Digium card. We ended up just re-seating the card and lo and behold, everything was hunky dory after re-seating. Does anyone know why? -- _ -- 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] 1.6.2 : global vars not read/set after #include w/ globals
Op 11-02-10 03:42, sean darcy schreef: Kevin P. Fleming wrote: sean darcy wrote: I found out that the [globals] section in extensions.conf is ignored if an #include 'd file has a [globals] section. Is this intended? In this particular case, the #include 'd file has a number of contexts for googlevoice. I'd put various googlevoice variables in there to use in all those contexts. Once I did that all of the global variables set in extensions.conf were ignored. Context names cannot be duplicated, unless you suffix them with (+) to allow them to be added together. It does not matter whether it is the 'global' context or any other context. Well Dialplan reloaded. == Parsing '/etc/asterisk/extensions.conf': == Found .. == Parsing '/etc/asterisk/exts/gvoice.exten.conf': == Found cat exts/gvoice.exten.conf [+globals] test-global = need-a-plus-sign . but no test-global in dialplan show globals :( sean suffix means 'append to the end'... so try [globals+] Ron -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer -- _ -- 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
