[asterisk-users] Dependencies for BETTER_BACKTRACES on Centos 5.6
Hi I'm struggling to find the dependencies to allow me to tick BETTER_BACKTRACES while installing asterisk 1.8.7 on CentOS 5.6 Does anyone know what I need to install to do this? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] vigor 2920 problems
One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video calls not working
Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten = 111,1,Answer() same = n,Dial(SIP/2206,60,r) same = n,Hangup() *SIP.conf* [2218] type=friend secret=*** callerid=Virendra 9172341457 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all dtmfmode=inband insecure=invite,port ;context=outbound context=bhati-test qualify=yes accountcode=123654789 disallow = all allow = ulaw,alaw,h263,g729,gsm,h264 videosupport=yes [2206] type=friend secret=*** callerid=2206 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all dtmfmode=inband insecure=invite,port context=outbound qualify=yes disallow = all allow = ulaw,alaw,h263,g729,gsm,h264 videosupport=yes *codec list of asterisk 1.6.2.11* *haddock8-astrx*CLI core show codecs* Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) 65536 (1 16) (0x1) image jpeg (JPEG image) 131072 (1 17) (0x2) imagepng (PNG image) 262144 (1 18) (0x4) video h261 (H.261 Video) 524288 (1 19) (0x8) video h263 (H.263 Video) 1048576 (1 20) (0x10) video h263p (H.263+ Video) 2097152 (1 21) (0x20) video h264 (H.264 Video) haddock8-astrx*CLI *CLI Output:-* -- Executing [111@bhati-test:1] Answer(SIP/2218-0664, ) in new stack -- Executing [111@bhati-test:2] Dial(SIP/2218-0664, SIP/2206,60,r) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 2206 -- SIP/2206-0665 is ringing -- SIP/2206-0665 is ringing [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' -- SIP/2206-0665 answered SIP/2218-0664 [Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:30] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:34] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:40] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:44] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:50] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:54] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:58:00] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:58:04] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:58:11] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:58:13] NOTICE[7924]: chan_sip.c:21479 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 2206 [Nov 21 15:58:15] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:58:21] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126
Re: [asterisk-users] vigor 2920 problems
Hi John, We've had similiar issues with customers behind the 2920 connecting to a hosted asterisk system. If you rebooted a phone it often didn't re-register, Checking the NAT sessions table on the router revealed stale nat sessions open for the phone. On the advice of Dreytek we found a fix by lowering the NAT session timeout from the default of 24hrs down to 5 minutes and installing the latest release of the firmware (3.3.7) it may not be available on the UK Site at the moment (It wasn't when we did the upgrade!) but it can be got from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/ It may help, It may not - But its quick easy fix if it does. Regards, AJ. - Original Message - From: John Taylor j...@vetsurgeon.org.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 21 November, 2011 10:20:14 AM Subject: [asterisk-users] vigor 2920 problems One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
If the caller hangs up Asterisk sends a SIGHUP. You can catch the signal and do whatever you want to do. Am 21.11.2011 07:38, schrieb David Cunningham: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thorsten Gllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Dsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
We do that with the F option in Dial(). From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial : F(context^exten^pri): When the caller hangs up, transfer the called party to the specified context and extension and continue execution. Cheers, Kingsley. On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files and spool directiory shared amongst several asterisk servers
As far as I know the linux kernel uses inotify to give Asterisk a hint, that a new call file is available. Does inotify work in your environment (external storage device) at all? Am 18.11.2011 11:29, schrieb Ishfaq Malik: We have a number of asterisk servers that share a spool directory on an external storage device (for call recording). We don't use call files but now are about to just purely for our own reporting purposes. Has anyone got any experience on the behaviour of using call files when several asterisk servers share a single spool directory? We are using 1.8 Thanks Ish -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vigor 2920 problems
Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will get permission to try new firmware later! JT On 21 November 2011 10:45, Arthur Stanfield a...@dmcip.com wrote: Hi John, We've had similiar issues with customers behind the 2920 connecting to a hosted asterisk system. If you rebooted a phone it often didn't re-register, Checking the NAT sessions table on the router revealed stale nat sessions open for the phone. On the advice of Dreytek we found a fix by lowering the NAT session timeout from the default of 24hrs down to 5 minutes and installing the latest release of the firmware (3.3.7) it may not be available on the UK Site at the moment (It wasn't when we did the upgrade!) but it can be got from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/ It may help, It may not - But its quick easy fix if it does. Regards, AJ. - Original Message - From: John Taylor j...@vetsurgeon.org.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 21 November, 2011 10:20:14 AM Subject: [asterisk-users] vigor 2920 problems One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] check if devices reachable in queue
On 11/20/2011 02:49 PM, Matt Hamilton wrote:
2. if the devices/members in the queue are not reachable, I would like
to forward him to a phone B.
I'm looking for a fast/practical way of accomplishing the second one.
In other words, before sending a call to a queue, I would like to see
if the members/devices in that queue are available/reachable.
I define the members statically in queue.conf and QUEUE_MEMBER_COUNT
gives the count of those - doesn't care if they are
available/reachable or not (even if phone is unhooked, still counted).
I should be able to loop through each member and use
${DEVICE_STATE(sip phone)}. for every incoming call, isn't this
overkill? Any other way?
Have you tried, instead of pre-processing the caller before calling
Queue(), checking the ${QUEUESTATUS} variable. On a Timeout, it will be
TIMEOUT, but there are also JOINEMPTY, LEAVEEMPTY, JOINEUNAVAIL and
LEAVEUNAVAIL options as well (core show application Queue). If you set
your queues.conf to consider the queue to be empty when the members are
unavailable, invalid or unknown, the Queue() app should return
immediately to the next dialplan step with the QUEUSTATUS of JOINEMPTY.
Dale
--
The truth speaks for itself. I'm just the messenger.
Lyta Alexander - Babylon 5
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Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Thorsten, We have SIGHUP set to 'IGNORE', but it still does not continue the AGI after the Dial(). Do you have any idea why that might happen? Thanks for your advice. On 21 November 2011 22:19, Thorsten Göllner t...@ovm-group.com wrote: If the caller hangs up Asterisk sends a SIGHUP. You can catch the signal and do whatever you want to do. Am 21.11.2011 07:38, schrieb David Cunningham: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Kingsley, Thanks for the reply, but I am looking to continue within the same AGI process and I believe that method would require starting a new AGI. On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk wrote: We do that with the F option in Dial(). From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial : F(context^exten^pri): When the caller hangs up, transfer the called party to the specified context and extension and continue execution. Cheers, Kingsley. On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
hello, try to delete all spaces between user and password on the pass.txt Regards - Bakko-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR mysql with asterisk 1.4
hello list i have asterisk 1.4 installed and i want to use CDR mysql during the installation i didn’t check the cdr mysql with make menuselect my question : i want to check this option now after the installtion and configuration of all options but he asks me to do. /configure before to use make menuselect i want to know if there any problem if i do. / configure and make menuselect to install cdr because this server is very important for me and i can’t stop it thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
Hi, After deleting all space no improvements. On Mon, Nov 21, 2011 at 5:35 PM, bakko asannu...@gmail.com wrote: ** hello, try to delete all spaces between user and password on the pass.txt Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR mysql with asterisk 1.4
salaheddine elharit wrote: because this server is very important for me and i can’t stop it This is one of those schedule for after hours things then. I don't believe you can do this without a restart of the Asterisk service. But, down time should be minimal. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Hi,
I use an AGI with PHP. Here is a short snippet:
[...]
declare(ticks = 1);
pcntl_signal(SIGHUP, array($this, "signal_handler"));
[...]
public function signal_handler($signal_number)
{
$this-log_message("debug", "Signal catched:
signo=$signal_number");
switch($signal_number) {
case SIGHUP:
// signal for hangup (comes from asterisk)
$this-log_message("debug", "Hangup
detected.");
exit(0);
break;
default:
$this-log_message("error", "Undefined signal
'".$signal_number."'.");
break;
}
}
Work for me. Give it a try.
Best regards,
-Thorsten-
Am 21.11.2011 13:00, schrieb David Cunningham:
Thorsten,
We have SIGHUP set to 'IGNORE', but it still does not continue the
AGI after the Dial(). Do you have any idea why that might happen?
Thanks for your advice.
On 21 November 2011 22:19, Thorsten Gllner t...@ovm-group.com
wrote:
If the caller hangs up
Asterisk sends a SIGHUP. You can catch the signal and do
whatever you want to do.
Am 21.11.2011 07:38, schrieb David Cunningham:
Hello,
We would like to continue a Perl AGI after a Dial() it
has done completes following caller hangup. We would
like to do this in the same AGI, and not using a new
AGI from the 'h' extension. It works fine when the
called party hangs up and the 'g' option is used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be
very helpful.
Thanks for any advice!
--
Thorsten Gllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Dsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-users] CDR mysql with asterisk 1.4
On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit salah.elharit...@gmail.com wrote: hello list i have asterisk 1.4 installed and i want to use CDR mysql during the installation i didn’t check the cdr mysql with make menuselect my question : i want to check this option now after the installtion and configuration of all options but he asks me to do. /configure before to use make menuselect i want to know if there any problem if i do. / configure and make menuselect to install cdr because this server is very important for me and i can’t stop it How did you initially install Asterisk? When compiling from source ./configure is the first step before you can run make. It shouldn't prompt to run ./configure for make menuselect if you are just changing some options from a previously compile and install. If you were able to run make menuselect without configure you might be able to load the module while Asterisk is running. You would copy the cdr_mysql.so to the lib directory and run module load cdr_mysql. However I would still plan this for after hours in case of an issue. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Yeah I think I slightly misread your original question, which I realised when I saw Thorsten's reply. I initially thought you just wanted to avoid going into the h extension. I'm not doing any AGI stuff here that hangs around while the call does stuff - the AGI process just runs quickly then quits, returning control back to the dialplan. I had incorrectly assumed you were doing the same. Cheers, Kingsley. On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote: Kingsley, Thanks for the reply, but I am looking to continue within the same AGI process and I believe that method would require starting a new AGI. On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk wrote: We do that with the F option in Dial(). From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial : F(context^exten^pri): When the caller hangs up, transfer the called party to the specified context and extension and continue execution. Cheers, Kingsley. On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Read() application
I tried to patch app_read on my development dahdi box as follows:
static int unload_module(void)
{
int res;
res = ast_unregister_application(app);
/* ast_module_user_hangup_all(); */
return res;
}
But the offending behavior persists - it's not a show-stopper but it
eventually could be.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Saturday, November 19, 2011 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Read() application
Hi,
Did you get a workaround for this? I sent you a message offlist but you
didn't reply so I don't know whether you saw it.
Cheers,
Kingsley.
On Fri, 2011-11-18 at 13:15 -0600, Danny Nicholas wrote:
My IVR wouldn't sound right if I allowed 2 or 3 times before it was
considered a failure. The big(ger) problem is that it just hangs up when
it fails, no warning or work around to do.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug
Lytle
Sent: Friday, November 18, 2011 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Read() application
Danny Nicholas wrote:
The user reported to me that I punched 1 and it hung up - in my
testing,
I found that slow DTMF entry (1 digit every 2 seconds or so) or fast
entry (more than 10 digits per second) was most likely to cause the
problem.
I've never had mine just hangup on a mis-key, but then again I have it
try 3 times before considering it a failure.
exten = s,1,Read(get-admin-password|enter-password|||3|)
Doug
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Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Just offhand, I think you should utilize the FastAGI protocol, since it
doesn't seem to live or die based on when the call hangs up. Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial() following caller
hang up?
Yeah I think I slightly misread your original question, which I realised
when I saw Thorsten's reply. I initially thought you just wanted to avoid
going into the h extension.
I'm not doing any AGI stuff here that hangs around while the call does stuff
- the AGI process just runs quickly then quits, returning control back to
the dialplan. I had incorrectly assumed you were doing the same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within the same AGI
process and I believe that method would require starting a new AGI.
On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the
called
party to the specified context and extension and continue
execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote:
Hello,
We would like to continue a Perl AGI after a Dial() it has
done
completes following caller hangup. We would like to do this
in the
same AGI, and not using a new AGI from the 'h' extension. It
works
fine when the called party hangs up and the 'g' option is
used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be very
helpful.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
--
_
-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
every Thurs:
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_
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Thurs:
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--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
Cheers,
Kingsley.
--
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Re: [asterisk-users] CDR mysql with asterisk 1.4
i try to run make menuselect without configure but he give me an error and he tell me that i must run ./configure before launch make menuselect i'm afraid if i launch ./configure and after make menuselet to lost all configuration related to asterisk BTW i can restart asterisk without issue thanks for your response 2011/11/21 Ryan Wagoner rswago...@gmail.com On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit salah.elharit...@gmail.com wrote: hello list i have asterisk 1.4 installed and i want to use CDR mysql during the installation i didn’t check the cdr mysql with make menuselect my question : i want to check this option now after the installtion and configuration of all options but he asks me to do. /configure before to use make menuselect i want to know if there any problem if i do. / configure and make menuselect to install cdr because this server is very important for me and i can’t stop it How did you initially install Asterisk? When compiling from source ./configure is the first step before you can run make. It shouldn't prompt to run ./configure for make menuselect if you are just changing some options from a previously compile and install. If you were able to run make menuselect without configure you might be able to load the module while Asterisk is running. You would copy the cdr_mysql.so to the lib directory and run module load cdr_mysql. However I would still plan this for after hours in case of an issue. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR mysql with asterisk 1.4
From what I read you are running a pre-compiled asterisk - what you can do in that instance is this 1 create a directory like /usr/local/src/asterisk/1.4-update 2 wget the matching version as indicated by core show version 3 extract the tar to the directory from step 1 4 run ./configure 5 run make menuselect 6 run make - DO NOT RUN make install 7 copy cdr_mysql.so to /usr/lib/asterisk/modules and load the module from CLI From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Monday, November 21, 2011 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR mysql with asterisk 1.4 i try to run make menuselect without configure but he give me an error and he tell me that i must run ./configure before launch make menuselect i'm afraid if i launch ./configure and after make menuselet to lost all configuration related to asterisk BTW i can restart asterisk without issue thanks for your response 2011/11/21 Ryan Wagoner rswago...@gmail.com On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit salah.elharit...@gmail.com wrote: hello list i have asterisk 1.4 installed and i want to use CDR mysql during the installation i didn't check the cdr mysql with make menuselect my question : i want to check this option now after the installtion and configuration of all options but he asks me to do. /configure before to use make menuselect i want to know if there any problem if i do. / configure and make menuselect to install cdr because this server is very important for me and i can't stop it How did you initially install Asterisk? When compiling from source ./configure is the first step before you can run make. It shouldn't prompt to run ./configure for make menuselect if you are just changing some options from a previously compile and install. If you were able to run make menuselect without configure you might be able to load the module while Asterisk is running. You would copy the cdr_mysql.so to the lib directory and run module load cdr_mysql. However I would still plan this for after hours in case of an issue. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video calls not working
Two items #1 you only need 1 disallow=all in your sip.conf definition #2 you need to patch rtp.c to define 126 as FORMAT_H263 - this is an xlite response to Asterisk starting music-on-hold during the connect pause. The r on the dial command attempts to do a faux ring which xlite interprets as a MOH request, so if you don't want to patch/recompile, just take the r off of Dial. From: virendra bhati [mailto:virbh...@gmail.com] Sent: Monday, November 21, 2011 4:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Sam Govind; Danny Nicholas Subject: video calls not working Hi list, I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration. Extensions.conf exten = 111,1,Answer() same = n,Dial(SIP/2206,60,r) same = n,Hangup() SIP.conf [2218] type=friend secret=*** callerid=Virendra 9172341457 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all dtmfmode=inband insecure=invite,port ;context=outbound context=bhati-test qualify=yes accountcode=123654789 disallow = all allow = ulaw,alaw,h263,g729,gsm,h264 videosupport=yes [2206] type=friend secret=*** callerid=2206 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all dtmfmode=inband insecure=invite,port context=outbound qualify=yes disallow = all allow = ulaw,alaw,h263,g729,gsm,h264 videosupport=yes codec list of asterisk 1.6.2.11 haddock8-astrx*CLI core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) 65536 (1 16) (0x1) image jpeg (JPEG image) 131072 (1 17) (0x2) imagepng (PNG image) 262144 (1 18) (0x4) video h261 (H.261 Video) 524288 (1 19) (0x8) video h263 (H.263 Video) 1048576 (1 20) (0x10) video h263p (H.263+ Video) 2097152 (1 21) (0x20) video h264 (H.264 Video) haddock8-astrx*CLI CLI Output:- -- Executing [111@bhati-test:1] Answer(SIP/2218-0664, ) in new stack -- Executing [111@bhati-test:2] Dial(SIP/2218-0664, SIP/2206,60,r) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 2206 -- SIP/2206-0665 is ringing -- SIP/2206-0665 is ringing [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' -- SIP/2206-0665 answered SIP/2218-0664 [Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:30] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:34] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:40] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:44] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:57:50] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:54] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191' [Nov 21 15:58:00] NOTICE[30518]: rtp.c:1811
[asterisk-users] queue ring delay
Hi, Does a parameter exist for a queue to delay ringing/sending a caller to all agent phones after the previous call is answered by an agent? My queue ring strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones. And it looks like the KIRK wireless server may need a split send to realize all wireless phones are no longer ringing (busy) after 1 call rings is unanswered, prior to sending a 2nd call. In other words, I think that what we are currently experiencing is this: Incoming call gets routed to our queue. It rings all phones. In the meantime, a 2nd caller gets routed to our queue (in line behing the first caller that is currently ringing our phones). One queue agent answers the first phone call in the queue. Asterisk immediately starts ringing all queue agent extensions again with the 2nd caller. However, most of the agents extensions are reporting busy, and so their phones don't ring. The 2nd caller may wind up getting routed to our queue fallback destination. So it seems to me that asterisk is sending the 2nd call to the queue agents before their phones are ready (i.e. before the KIRK Wireless Server is able to realize that they are no longer ringing [busy] from the first caller). So I'm thinking that if I can introduce some type of delay of 500ms-1 second AFTER a queue call rings all phones, but before a subsequent call is permitted to ring all phones, my problem will be solved. I am using FreePBX. I know this is not the place to get FreePBX support, but I believe that the FreePBX gui is just providing a front-end for standard asterisk features parameters behind the scenes. I am on Digium AsteriskNOW with asterisk 1.6. I also believe that this mailing list may be the best source of community support for asterisk, so I am posting here. :-) From within FreePBX, in the queue configuration, I have a parameter for Wrap-Up-Time and Member Delay. My questions would be: 1. Does Wrap-Up-Time apply to all queue agents/extensions that just rang, or only the one who actually answered the call (I assume the latter)? 2. Does the Member Delay delay the ringing of new calls to agents, or only come into play AFTER the agent answers the ringing call? Any other suggestions for how I can resolve this issue? I am wondering whether Agent Timeout or Agent Timeout Restart (or a combination of both) may be able to help me here. It sounds like the 2nd option may help me. But I'm not familiar with exactly how it would work in this situation. Anyway, that's it. As for some background, we initially were using ring groups, but realized that these phones do NOT have the ability to handle a 2nd ringing call. So in the event that 2 inbound calls rang within a few seconds, asterisk would send the first to all phones, and then when tyring to send the 2nd, would receive a BUSY message from the phones (because they were busy processing a ring for the first caller), and the 2nd caller would wind up going straight to the unavilable destination for the ring group, instead of eventually ringing through to the phones after someone answered the first call. I greatly appreciate your help insight with this issue! - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.comhttp://www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting AstDB family at start
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Sunday, November 20, 2011 1:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Deleting AstDB family at start Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. I know there is a DBdeltree(family) function. Is there a context that only runs once (automatically) at Asterisk startup (so that I can call this function)? Also is AstDB lookup faster than a func_odbc lookup? Is there a faster way to perform a lookup in Asterisk; e.g. create a lookup table in memory perhaps? I'm new to Asterisk... Thanks, Matt After a little research and testing, these are my answers: Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. not explicitly. The [general] context is executed at asterisk reload (answer to question 2), but doesn't execute dialplan functions. You could put a call in /usr/sbin/safe_asterisk to clear your keys using a local call in /var/spool/asterisk/outgoing [clearkeys] Exten = start,1,answer() Exten = start,n,dbdeltree(foo) Exten = start,n,hangup Also is AstDB lookup faster than a func_odbc lookup? IMO yes since Asterisk has a built-in connection to it's Berkley or SQLite database and the odbc lookup has to go through more layers. Finally Is there a faster way to perform a lookup in Asterisk; e.g. create a lookup table in memory perhaps? Set and retrieve Global variables for small searches. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no more trunk with asterisk
hello list i have asterisk 1.4 with and i have one card diguim (E1) with 2 providers i have noticed by an error related to the first provider some times i can not call the numbers of of this provider but with the second one there is no issue alos i cal call the internal extension without issue for exemple today between 15h55 and 16h10 i have noticed this issue if you can see any thing in this log and tell me what is the problem for your information i restart asterisk in order to solve this issue all the time thanks and regards [Nov 21 15:55:02] DEBUG[29762] chan_zap.c: Queuing frame from PRI_EVENT_PROGRESS on channel 0/19 span 2 [Nov 21 15:55:04] NOTICE[30280] ThreadPoolMngr.c: Got a thread from the pool [1968421792] [Nov 21 15:55:04] NOTICE[30280] ThreadPoolMngr.c: Thread [1968421792] signaled [Nov 21 15:55:04] DEBUG[29761] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 1/4 span 1 [Nov 21 15:55:04] DEBUG[30379] app_ahagentdistribute.c: Timeout not activated for AhAgentDistribute [Nov 21 15:55:04] DEBUG[30379] app_ahagentdistribute.c: Call on channel [Local/OUT0660108860@aheeva_ccs-bde4,1] need to be queued until the CTI sends an agent to distribute [Nov 21 15:55:04] DEBUG[30379] app_ahEventsProxy.c: addRequest: - chan-uniqueid[1321890889.169677] [Nov 21 15:55:04] DEBUG[30379] app_ahEventsProxy.c: AHGetQueueAddress: Track Number for channel[Local/OUT0660108860@aheeva_ccs-bde4,1] is 1321890889.198883 [Nov 21 15:55:04] DEBUG[30379] app_ahEventsProxy.c: AHGetQueueAddress: Response from Aheeva CTI [local^1011^] [Nov 21 15:55:04] DEBUG[30379] app_ahEventsProxy.c: AHGetQueueAddress: agent's asterisk ip[local] port[(null)] and agent id[1011] [Nov 21 15:55:04] DEBUG[30379] app_AheevaPhoneSimulator.c: client is not an aheeva phone, so using default values. [Nov 21 15:55:05] DEBUG[29761] chan_zap.c: Echo cancellation already on [Nov 21 15:55:05] DEBUG[30377] app_ahagentdistribute.c: Timeout not activated for AhAgentDistribute [Nov 21 15:55:05] DEBUG[30377] app_ahagentdistribute.c: Call on channel [Local/OUT0613933851@aheeva_ccs-c3c8,1] need to be queued until the CTI sends an agent to distribute [Nov 21 15:55:05] DEBUG[30377] app_ahEventsProxy.c: addRequest: - chan-uniqueid[1321890894.169683] [Nov 21 15:55:05] DEBUG[30377] app_ahEventsProxy.c: AHGetQueueAddress: Track Number for channel[Local/OUT0613933851@aheeva_ccs-c3c8,1] is 1321890894.198885 [Nov 21 15:55:05] DEBUG[30377] app_ahEventsProxy.c: AHGetQueueAddress: Response from Aheeva CTI [local^1015^] [Nov 21 15:55:05] DEBUG[30377] app_ahEventsProxy.c: AHGetQueueAddress: agent's asterisk ip[local] port[(null)] and agent id[1015] [Nov 21 15:55:05] DEBUG[30377] app_AheevaPhoneSimulator.c: client is not an aheeva phone, so using default values. [Nov 21 15:55:06] DEBUG[30379] chan_agent.c: Bridge on 'IAX2/1011-9' being set to 'Agent/1011' (3) [Nov 21 15:55:07] DEBUG[30377] chan_agent.c: Bridge on 'IAX2/1015-10' being set to 'Agent/1015' (3) [Nov 21 15:55:08] DEBUG[29761] chan_zap.c: Echo cancellation already on [Nov 21 15:55:08] DEBUG[30380] app_ahagentdistribute.c: Timeout not activated for AhAgentDistribute [Nov 21 15:55:08] DEBUG[30380] app_ahagentdistribute.c: Call on channel [Local/OUT0671825124@aheeva_ccs-6ad8,1] need to be queued until the CTI sends an agent to distribute [Nov 21 15:55:08] DEBUG[30380] app_ahEventsProxy.c: addRequest: - chan-uniqueid[1321890889.169675] [Nov 21 15:55:08] DEBUG[30380] app_ahEventsProxy.c: AHGetQueueAddress: Track Number for channel[Local/OUT0671825124@aheeva_ccs-6ad8,1] is 1321890889.198882 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR mysql with asterisk 1.4
thanks danny and thanks all i will test this solution and i will update you by the result kind regards 2011/11/21 Danny Nicholas da...@debsinc.com From what I read you are running a pre-compiled asterisk – what you can do in that instance is this 1 create a directory like /usr/local/src/asterisk/1.4-update 2 wget the matching version as indicated by “core show version” 3 extract the tar to the directory from step 1 4 run ./configure 5 run make menuselect 6 run make – DO NOT RUN make install 7 copy cdr_mysql.so to /usr/lib/asterisk/modules and load the module from CLI ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Monday, November 21, 2011 8:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CDR mysql with asterisk 1.4 ** ** i try to run make menuselect without configure but he give me an error and he tell me that i must run ./configure before launch make menuselect * *** i'm afraid if i launch ./configure and after make menuselet to lost all configuration related to asterisk BTW i can restart asterisk without issue thanks for your response 2011/11/21 Ryan Wagoner rswago...@gmail.com On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit salah.elharit...@gmail.com wrote: hello list i have asterisk 1.4 installed and i want to use CDR mysql during the installation i didn’t check the cdr mysql with make menuselect my question : i want to check this option now after the installtion and configuration of all options but he asks me to do. /configure before to use make menuselect i want to know if there any problem if i do. / configure and make menuselect to install cdr because this server is very important for me and i can’t stop it How did you initially install Asterisk? When compiling from source ./configure is the first step before you can run make. It shouldn't prompt to run ./configure for make menuselect if you are just changing some options from a previously compile and install. If you were able to run make menuselect without configure you might be able to load the module while Asterisk is running. You would copy the cdr_mysql.so to the lib directory and run module load cdr_mysql. However I would still plan this for after hours in case of an issue. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR uniqueid - across multiple servers?
Hi, Is there a way to add a uniqueid prefix to each server to make sure that the CDRs uniqueids are indeed unique across multiple servers? I am using MYSQL tables to keep these records. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] difference between playback and background?
In the dial plan language of asterisk, what is the difference between prompting
the user with a Playback() command vs. a Background() command? I want in a part
of my dial plan to ask the user a prompt, and wait for 4 digits to be typed in.
I don't want the user to have to end the string with a pound or something, just
wait 2 seconds after they stop typing. ANd I do want the prompt to be
interruptible if the user is fast and knows already what to do…
I need to do some tests on the number they entered. If i use background(), and
say the prompt, and then follow with a WAIT command, how do i reference the
number they just typed in? does asterisk set the ${EXTEN} variable when the
user types something?
What I find maddening about the asterisk documentation is a lack of clarity on
the sequence of things, and what variables get set when?
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Re: [asterisk-users] queue ring delay
So I found a good description of the timeoutrestart setting here https://issues.asterisk.org/view.php?id=12690#87263. It definitely isn't what I'm looking for. So I think I may be left with two options: 1. Set Skip Busy Agents to No. (not sure how this will work with my KIRK phones. Currently I have call-waiting disabled on these phones, as they are not intuitive for handling a 2nd call while already on the phone with 1 call. So I'm not sure whether asterisk would continue to try to send the queue calls these phones (during this split second while the phones are still reporting a status of Ringing/BUSY), or whether it would actually send the call to the extension's VM (which would be even worse).. 2. Manually adjust the diaplan to introduce some delay after a ringing queue call is answered by an agent, but before the subsequent call ring the queue agents. If this becomes the solution, I may need some assistance (although I'm sure I'd eventually figure it out). Again, your help is appreciated. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Douglas Mortensen [mailto:d...@impalanetworks.com] Sent: Monday, November 21, 2011 9:56 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] queue ring delay Hi, Does a parameter exist for a queue to delay ringing/sending a caller to all agent phones after the previous call is answered by an agent? My queue ring strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones. And it looks like the KIRK wireless server may need a split send to realize all wireless phones are no longer ringing (busy) after 1 call rings is unanswered, prior to sending a 2nd call. In other words, I think that what we are currently experiencing is this: Incoming call gets routed to our queue. It rings all phones. In the meantime, a 2nd caller gets routed to our queue (in line behing the first caller that is currently ringing our phones). One queue agent answers the first phone call in the queue. Asterisk immediately starts ringing all queue agent extensions again with the 2nd caller. However, most of the agents extensions are reporting busy, and so their phones don't ring. The 2nd caller may wind up getting routed to our queue fallback destination. So it seems to me that asterisk is sending the 2nd call to the queue agents before their phones are ready (i.e. before the KIRK Wireless Server is able to realize that they are no longer ringing [busy] from the first caller). So I'm thinking that if I can introduce some type of delay of 500ms-1 second AFTER a queue call rings all phones, but before a subsequent call is permitted to ring all phones, my problem will be solved. I am using FreePBX. I know this is not the place to get FreePBX support, but I believe that the FreePBX gui is just providing a front-end for standard asterisk features parameters behind the scenes. I am on Digium AsteriskNOW with asterisk 1.6. I also believe that this mailing list may be the best source of community support for asterisk, so I am posting here. :-) From within FreePBX, in the queue configuration, I have a parameter for Wrap-Up-Time and Member Delay. My questions would be: 1. Does Wrap-Up-Time apply to all queue agents/extensions that just rang, or only the one who actually answered the call (I assume the latter)? 2. Does the Member Delay delay the ringing of new calls to agents, or only come into play AFTER the agent answers the ringing call? Any other suggestions for how I can resolve this issue? I am wondering whether Agent Timeout or Agent Timeout Restart (or a combination of both) may be able to help me here. It sounds like the 2nd option may help me. But I'm not familiar with exactly how it would work in this situation. Anyway, that's it. As for some background, we initially were using ring groups, but realized that these phones do NOT have the ability to handle a 2nd ringing call. So in the event that 2 inbound calls rang within a few seconds, asterisk would send the first to all phones, and then when tyring to send the 2nd, would receive a BUSY message from the phones (because they were busy processing a ring for the first caller), and the 2nd caller would wind up going straight to the unavilable destination for the ring group, instead of eventually ringing through to the phones after someone answered the first call. I greatly appreciate your help insight with this issue! - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.comhttp://www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options
Re: [asterisk-users] difference between playback and background?
It sounds like you may want to use the READ command instead. This lets you
hard-set the number of digits to expect and then sets a variable which you
can use later in the dialplan. Generally you use the background command to
let them dial an extension or automated attendant option. Playback plays
without the option to interrupt it.
On Mon, Nov 21, 2011 at 10:50 AM, Edward de Jong
edward.dej...@voicecarrier.com wrote:
In the dial plan language of asterisk, what is the difference between
prompting the user with a Playback() command vs. a Background() command? I
want in a part of my dial plan to ask the user a prompt, and wait for 4
digits to be typed in. I don't want the user to have to end the string with
a pound or something, just wait 2 seconds after they stop typing. ANd I do
want the prompt to be interruptible if the user is fast and knows already
what to do…
I need to do some tests on the number they entered. If i use background(),
and say the prompt, and then follow with a WAIT command, how do i reference
the number they just typed in? does asterisk set the ${EXTEN} variable when
the user types something?
What I find maddening about the asterisk documentation is a lack of
clarity on the sequence of things, and what variables get set when?
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Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] difference between playback and background?
First question - playback is not interruptable by DTMF, background is.
You have two options here
Option 1
Use Read
[getnum]
Exten = start,1,read(mydigit,prompt,4,skip,1,2)
.. verification stuff
Option 2
Use WaitExten with Background
[getnum]
Exten = start,1,background(prompt)
Exten = start,n,waitexten(2)
Exten = ,1,noop(user pressed )
Exten = I,1,playback(invalid)
For option 2 you have to define each valid 4 digit entry in the context.
Yes it can be maddening, but you get what you pay for.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Edward de Jong
Sent: Monday, November 21, 2011 11:51 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] difference between playback and background?
In the dial plan language of asterisk, what is the difference between
prompting the user with a Playback() command vs. a Background() command? I
want in a part of my dial plan to ask the user a prompt, and wait for 4
digits to be typed in. I don't want the user to have to end the string with
a pound or something, just wait 2 seconds after they stop typing. ANd I do
want the prompt to be interruptible if the user is fast and knows already
what to do.
I need to do some tests on the number they entered. If i use background(),
and say the prompt, and then follow with a WAIT command, how do i reference
the number they just typed in? does asterisk set the ${EXTEN} variable when
the user types something?
What I find maddening about the asterisk documentation is a lack of clarity
on the sequence of things, and what variables get set when?
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Re: [asterisk-users] Problem with Atxfer for the calling party
Hi There,
I'm still having this problem, Does somebody know what can be
happening?
Regards.
On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
Hello,
The exten is the parameter passed to the macro, which contains the
sip device name. I'll change the name to another less confusing.
* Alexandre, também sou brasileiro hehe, notei que você já escreveu um
livro sobre asterisk, será que você poderia me ajudar com esse
problema? Já tem alguns dias que estou na luta aqui hehe.
On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller wrote:
You're using ${exten} inside your macro, you should use ${EXTEN}.
--
Atenciosamente,
ALEXANDRE KELLER
http://twitter.com/alexandrekeller
http://www.facebook.com/alexandre.keller.BR
Dinheiro é a consequência de um trabalho bem feito e não o motivo
para se fazer um bom trabalho.
P Antes de imprimir pense em seu compromisso com o Meio Ambiente.
On 11/11/2011, at 08:38, Antonio Modesto wrote:
On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas wrote:
It can have to do with either the telephones dial plan or the
context in the Asterisk dial plan combined with your
features.conf settings.
I noticed that my problem occurs when i use a macro to dial sip
devices, my dialplan is like this:
- Each sip device has its own context
- This context includes the outgoing call contexts that this
extension can use for making calls and includes a context called
ramais, which has the dial plan to call another extensions, it
uses a macro to do this.
Here is the configuration for my extension modesto :
# sip.conf
[modesto](default_extension)
username=modesto
context=modesto
callerid=modesto 106
callgroup=4
pickupgroup=4
# Default extension template
type=friend
dtmfmode=auto
host=dynamic
disallow=all
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
canreinvite=yes
qualify=no
callcounter=yes
# context for SIP/modesto
context modesto {
includes {
vivo;
tim;
oi;
claro;
vivoddd;
timddd;
oiddd;
claroddd;
embratel;
embratel2;
};
includes {
ramais;
};
};
# Although the problem is occurring also for others contexts
included, i'll show only the ramais context, which is used to
call local extensions:
context ramais {
101 = dial_sip(suporte1);
102 = dial_sip(suporte2);
103 = dial_sip(suporte3);
105 = dial_sip(suporte05);
106 = dial_sip(modesto);
107 = dial_sip(gustavo);
108 = dial_sip(pauloh);
109 = dial_sip(fernanda);
111 = dial_sip(marcos);
112 = dial_sip(thiago);
115 = dial_sip(helder);
116 = dial_sip(atendimento01);
117 = dial_sip(atendimento03);
118 = dial_sip(atendimento02);
119 = dial_sip(marlon);
120 = dial_sip(suporteemp);
122 = dial_sip(telemais);
123 = dial_sip(casagustavo);
127 = dial_sip(manutencao);
128 = dial_sip(guilherme);
129 = dial_sip(marcelo);
130 = dial_sip(rafael);
132 = dial_sip(netita2);
133 = dial_sip(unotel);
};
If I use the Dial() application instead of this macro, it works
well. I noticed that when I use the macro and try to transfer a
call (The problem occurs only for the calling party, the called
party can do transfers with no problems), asterisk tries to find
the extension in the macro-name context and of course, there is
no dialplan to call the extensions there.
Here is the dial_sip macro:
macro dial_sip(exten) {
Verbose(2,== Chamando a MACRO dial_sip - ponto 1
macros.ael ==);
Verbose(4, Macro dial_sip iniciada.);
ChanIsAvail(SIP/${exten});
Verbose(2,== ${AVAILORIGCHAN});
if (${AVAILORIGCHAN} != )
{
Verbose(4, SIP/${exten} parece estar
disponivel, vou disca-lo agora.);
Set(FromExt=${CALLERID(num)});
System(/bin/sh /var/spool/asterisk/calllog/log.sh
SIP/${FromExt} SIP/${exten} SIP-TO-SIP);
Verbose(4, System status: ${SYSTEMSTATUS});
Dial(SIP/${exten},${SIP_DIAL_TIMEOUT},Ttr);
Hangup();
}
else
{
Verbose(2, SIP/${exten} nao esta
disponivel.);
Hangup();
};
NoOp(From ${MACRO_EXTEN} to ${exten});
System(${CALLLOGDIR}/log.sh ${exten});
Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Yeah fastAGI is great, I've been using it for a while for performance
reasons but yes I guess it would solve problems like this too.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
Just offhand, I think you should utilize the FastAGI protocol, since it
doesn't seem to live or die based on when the call hangs up. Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial() following caller
hang up?
Yeah I think I slightly misread your original question, which I realised
when I saw Thorsten's reply. I initially thought you just wanted to avoid
going into the h extension.
I'm not doing any AGI stuff here that hangs around while the call does stuff
- the AGI process just runs quickly then quits, returning control back to
the dialplan. I had incorrectly assumed you were doing the same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within the same AGI
process and I believe that method would require starting a new AGI.
On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the
called
party to the specified context and extension and continue
execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote:
Hello,
We would like to continue a Perl AGI after a Dial() it has
done
completes following caller hangup. We would like to do this
in the
same AGI, and not using a new AGI from the 'h' extension. It
works
fine when the called party hangs up and the 'g' option is
used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be very
helpful.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-users] Deleting AstDB family at start
Thanks Danny. [clearkeys] Exten = start,1,answer() Exten = start,n,dbdeltree(foo) Exten = start,n,hangup Set and retrieve Global variables for small searches. I will try the local call option to [clearkeys]. I guess I can also use a global flag to call dbdeltree only once in the existing context before entering anything into AstDB. Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR uniqueid - across multiple servers?
Since the MYSQL CDR is not the standard /var/log/asterisk/cdr-csv/Master.csv file, but an add_on where uniqueid is just a table field varchar(32), you could create an AGI to touch the field during the hangup extension and append the servername or a number to the front, so instead of 123456.111 you could have server1.123456.111 or you could make a daemon running outside of Asterisk to do the same thing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, November 21, 2011 11:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CDR uniqueid - across multiple servers? Hi, Is there a way to add a uniqueid prefix to each server to make sure that the CDRs uniqueids are indeed unique across multiple servers? I am using MYSQL tables to keep these records. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR uniqueid - across multiple servers?
Mike, Just enter a unique systemname into asterisk.conf for each box. This system identifier is appended to the front of the unique id field in cdr. /etc/asterisk/asterisk.conf [options] systemname=asterisk1 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 21 Nov 2011 13:50:26 -0600 Subject: Re: [asterisk-users] CDR uniqueid - across multiple servers? Since the MYSQL CDR is not the standard /var/log/asterisk/cdr-csv/Master.csv file, but an add_on where uniqueid is just a table field varchar(32), you could create an AGI to touch the field during the hangup extension and append the servername or a number to the front, so instead of 123456.111 you could have server1.123456.111 or you could make a daemon running outside of Asterisk to do the same thing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, November 21, 2011 11:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CDR uniqueid - across multiple servers? Hi, Is there a way to add a uniqueid prefix to each server to make sure that the CDRs uniqueids are indeed unique across multiple servers? I am using MYSQL tables to keep these records. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR uniqueid - across multiple servers?
Thank you, just what I was looking for. Danny: that`s a good solution, but I wanted something that didn't depend on one more extra script running. I have plenty of those already. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Monday, November 21, 2011 3:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDR uniqueid - across multiple servers? Mike, Just enter a unique systemname into asterisk.conf for each box. This system identifier is appended to the front of the unique id field in cdr. /etc/asterisk/asterisk.conf [options] systemname=asterisk1 _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 21 Nov 2011 13:50:26 -0600 Subject: Re: [asterisk-users] CDR uniqueid - across multiple servers? Since the MYSQL CDR is not the standard /var/log/asterisk/cdr-csv/Master.csv file, but an add_on where uniqueid is just a table field varchar(32), you could create an AGI to touch the field during the hangup extension and append the servername or a number to the front, so instead of 123456.111 you could have server1.123456.111 or you could make a daemon running outside of Asterisk to do the same thing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, November 21, 2011 11:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CDR uniqueid - across multiple servers? Hi, Is there a way to add a uniqueid prefix to each server to make sure that the CDRs uniqueids are indeed unique across multiple servers? I am using MYSQL tables to keep these records. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between playback and background?
On Mon, 21 Nov 2011, Danny Nicholas wrote: Option 2 Use WaitExten with Background [getnum] Exten = start,1,background(prompt) Exten = start,n,waitexten(2) Exten = ,1,noop(user pressed ) Exten = I,1,playback(invalid) For option 2 you have to define each valid 4 digit entry in the context. Or, (since the OP seems a bit newbish), read up on extension pattern matching. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting AstDB family at start
On Sun, 20 Nov 2011, Matt Hamilton wrote: Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. How about: [sudo] /usr/sbin/asterisk -r -x 'database deltree example' in /etc/init.d/asterisk or safe_asterisk? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting AstDB family at start
On 11-11-21 03:46 PM, Steve Edwards wrote: On Sun, 20 Nov 2011, Matt Hamilton wrote: Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. How about: [sudo] /usr/sbin/asterisk -r -x 'database deltree example' in /etc/init.d/asterisk or safe_asterisk? Easier to use cli.conf -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting AstDB family at start
What flavor does cli.conf start on? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Monday, November 21, 2011 3:34 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Deleting AstDB family at start On 11-11-21 03:46 PM, Steve Edwards wrote: On Sun, 20 Nov 2011, Matt Hamilton wrote: Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. How about: [sudo] /usr/sbin/asterisk -r -x 'database deltree example' in /etc/init.d/asterisk or safe_asterisk? Easier to use cli.conf -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEX410P drops DTMF digits
Hello again list, I'm running a 1.4.42 install on SUSE with an AEX410P card. The DAHDI release is 2.4.0 because the machine won't properly install 2.5 and also won't install Asterisk 10.0 because I can't get a good SQLite3 library to install. Whenever I enter DTMF very quickly or very slowly, app_read des on me. Has anyone experienced similar joy using DAHDI drivers? I've piddled with channel.c and app_read.c trying to tame this beast but it seems to have the better of me. Thanks in advance Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting AstDB family at start
Thanks Paul. The following works.. --cli.conf --- [startup_commands] ; ; Any commands listed in this section will get automatically executed ; when Asterisk starts as a daemon or foreground process (-c). ; ;sip set debug on = yes ;core set verbose 3 = yes ;core set debug 1 = yes database deltree example = yes --- Matt Date: Mon, 21 Nov 2011 16:33:47 -0500 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Deleting AstDB family at start On 11-11-21 03:46 PM, Steve Edwards wrote: On Sun, 20 Nov 2011, Matt Hamilton wrote: Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. How about: [sudo] /usr/sbin/asterisk -r -x 'database deltree example' in /etc/init.d/asterisk or safe_asterisk? Easier to use cli.conf -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] check if devices reachable in queue
Have you tried, instead of pre-processing the caller before calling
Queue(), checking the ${QUEUESTATUS} variable.
Even when the phones are UNREACHABLE, QUEUE is still trying until it times out
- ${QUEUESTATUS} = TIMEOUT
I get the following for all the members of the queue, in a loop, until it times
out.
Executing [1001@handle-queue:3] Dial(Local/1001@handle-queue-6d01;2,
SIP/1001) in new stack
[Nov 21 18:57:42] WARNING[4780]: app_dial.c:2196 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'Local/1001@handle-queue-6d01;2' status is
'CHANUNAVAIL'
-- Local/1001@handle-queue-6d01;1 is circuit-busy
-- Nobody picked up in 0 ms
[Nov 21 18:57:42] WARNING[4780]: channel.c:4622 ast_prod: Prodding channel
'Local/1001@handle-queue-6d01;2' failed
queue.conf-
joinempty=no
joinunavailable=no
leavewhenempty=yes
timeout=0(for testing purposes, I set the timeout in the
application to 10 secs)
timeoutpriority=app
timeoutrestart=no
retry=0
Is it possible to make the queue not wait for the timeout and return with
JOINUNAVAIL after 1 round of testing the peers?
Thanks.
Matt
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Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
The strange thing is that we are using fast AGI, and for some reason the
AGI always exits when the caller hangs up - even when I set HUP to IGNORE.
If I set HUP to a subroutine that just logs a message, that message is
never logged.
Thanks for all the help.
On 22 November 2011 05:23, Kingsley Tart kings...@skymarket.co.uk wrote:
Yeah fastAGI is great, I've been using it for a while for performance
reasons but yes I guess it would solve problems like this too.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
Just offhand, I think you should utilize the FastAGI protocol, since it
doesn't seem to live or die based on when the call hangs up. Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley
Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial() following caller
hang up?
Yeah I think I slightly misread your original question, which I realised
when I saw Thorsten's reply. I initially thought you just wanted to avoid
going into the h extension.
I'm not doing any AGI stuff here that hangs around while the call does
stuff
- the AGI process just runs quickly then quits, returning control back to
the dialplan. I had incorrectly assumed you were doing the same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within the same AGI
process and I believe that method would require starting a new AGI.
On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the
called
party to the specified context and extension and continue
execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote:
Hello,
We would like to continue a Perl AGI after a Dial() it has
done
completes following caller hangup. We would like to do this
in the
same AGI, and not using a new AGI from the 'h' extension. It
works
fine when the called party hangs up and the 'g' option is
used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be very
helpful.
Thanks for any advice!
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[asterisk-users] no sound with ICES ?
Hi, I'm trying to have Asterisk pick up a call and stream it to Liquidsoap (Icecast2 compatible). This is what I have in my extensions.conf : [default] exten = s,1,Answer exten = s,n,Wait(1) exten = s,n,Ices(/etc/asterisk/asterisk-ices.xml) exten = s,n,HangUp Here's what working so far: Asterisk does pick up the call, and connects to the icecast server. However it seems like nothing is sent in the stream, as if no sound were present. Doing Echo() instead of Ices() works fine (I can hear myself talk back). The ices binary is in the path, and works well with the supplied configuration file when called from the bash cli. Do you see anything that would explain that my voice is not sent through the Ices() function call ? Any help greatly appreciated Thomas-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resell VoIP Servcies
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