[asterisk-users] pjsip: outofcall_message_context
Hello. Is there an analog option outofcall_message_context for pjsip? or: how to determine that the call is an outbound text message? Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.2.0 Video issues
I can assure you that asterisk is crashing, as when I try to reconnect I see it reloading again. Could be that something is deleting the core ! is there a way to find the path to where the core files are stored? My system is Lubuntu , Linux #41 SMP PREEMPT Tue Nov 11 16:35:58 CST 2014 armv7l armv7l armv7l GNU/Linux Operating systemUbuntu Linux 14.04.1 --- Toufic KHREISH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Wednesday, March 18, 2015 4:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail) toufic.khre...@gmail.com wrote: I see that my asterisk is started with the -g option, the core file I cannot find on my system (find / -name core*) I would suspect one of the following: (1) Asterisk is not actually crashing. (2) Something is deleting the core files. (3) The core files are hiding really, really well. Either way, if you can't get a backtrace, there isn't much we can do to help with that problem. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] PRI Callerid Passthrough
In a word, no. PRI service providers will generally only allow the caller ID to be set to one of the numbers in the range that you have for inbound with them. On 18 Mar 2015 11:30, Rizwan H Qureshi rizwanhas...@gmail.com wrote: Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] PRI Callerid Passthrough
On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks It depends who your service provider is! Any PRI card can send the commands down the D-channel to set any caller ID you like, but it's still up to the telco whether or not they will honour your request. I know the hard way that BT will only let you identify with a number you're entitled to use. Also, remember if you have a call coming in on a PRI line and going out on another PRI line, that's eating two of your thirty lines . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.2.0 Video issues
If you take a look at the safe_asterisk shell script, usually located at /usr/sbin/safe_asterisk (for CentOS at least), you'll be able to find where the core files are located. If it's not located there, then you'll need to look at the Asterisk init script for the scripts location. I hope this helps. Regards; John -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Tuesday, March 17, 2015 11:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail) toufic.khre...@gmail.com wrote: I see that my asterisk is started with the -g option, the core file I cannot find on my system (find / -name core*) I would suspect one of the following: (1) Asterisk is not actually crashing. (2) Something is deleting the core files. (3) The core files are hiding really, really well. Either way, if you can't get a backtrace, there isn't much we can do to help with that problem. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no
On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome jl...@me.com wrote: Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table “direct_media” sets to “no” on all endpoints but it doesn’t help. if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it’s not a right way to fix this issues. A native_rtp bridge is used for more than direct media. It is also used for local native bridging, that is, when you have two RTP capable channels in a bridge and Asterisk does not require the media to flow through its core. The bridge then just performs a packet to packet swap between the two RTP capable channels. Note that on verbosity 4, Asterisk will tell you if the bridge is locally or remotely bridging the two channels. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
[asterisk-users] PRI Callerid Passthrough
Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] 4 Port PRI
Hi Guys I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes ;echocancelwhenbridged=no ;rxgain=0 ;txgain=0 callerid=asreceived musiconhold=default group=1 overlapdial=yes signalling=pri_cpe context=extensions channel = 1-15,17-31 jbenable= yes jbforce= yes jbmaxsize= 120 jbimpl= fixed jbresyncthreshold= 1000 -- _ -- 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] pjsip: outofcall_message_context
On Wed, Mar 18, 2015 at 4:43 AM, Dmitriy Serov serov@gmail.com wrote: Hello. Is there an analog option outofcall_message_context for pjsip? or: how to determine that the call is an outbound text message? The 'message_context' endpoint option [1] should provide what you're looking for. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_message_context -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] 4 Port PRI
I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes ;echocancelwhenbridged=no ;rxgain=0 ;txgain=0 callerid=asreceived musiconhold=default group=1 overlapdial=yes signalling=pri_cpe context=extensions channel = 1-15,17-31 jbenable= yes jbforce= yes jbmaxsize= 120 jbimpl= fixed jbresyncthreshold= 1000 PRI or BRI? Which card are you using? Typically the installation script or procedure lets you configure each span. You seem to have 4 spans for either 8 or 128 (EuroISDN) channels. jg -- _ -- 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] PRI Callerid Passthrough
Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough. On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks It depends who your service provider is! Any PRI card can send the commands down the D-channel to set any caller ID you like, but it's still up to the telco whether or not they will honour your request. I know the hard way that BT will only let you identify with a number you're entitled to use. Also, remember if you have a call coming in on a PRI line and going out on another PRI line, that's eating two of your thirty lines . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] 4 Port PRI
4 Port PRI sangoma a104 From: jg [mailto:webaccounts...@jgoettgens.de] Sent: Wednesday, March 18, 2015 2:09 PM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 4 Port PRI I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes ;echocancelwhenbridged=no ;rxgain=0 ;txgain=0 callerid=asreceived musiconhold=default group=1 overlapdial=yes signalling=pri_cpe context=extensions channel = 1-15,17-31 jbenable= yes jbforce= yes jbmaxsize= 120 jbimpl= fixed jbresyncthreshold= 1000 PRI or BRI? Which card are you using? Typically the installation script or procedure lets you configure each span. You seem to have 4 spans for either 8 or 128 (EuroISDN) channels. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no
Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table “direct_media” sets to “no” on all endpoints but it doesn’t help. if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it’s not a right way to fix this issues. Asterisk 13.2.0-- _ -- 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] 4 Port PRI
When parsing the config file, all the current settings are applied when the 'channel = ' directive is encountered. So something like this will make the three remaining groups and set signalling on ports 1 3 as pri_cpe and ports 2 4 as pri_net. ; setting specific to Group 2 group=2 signalling=pri_net channel = channels for group 2 ; settings specific to group 3 group=3 signalling=pri_cpe channel = channels for group 3 ; settings specific to group 4 group=4 signalling=pri_net channel = channels for group 4 BTW, this also means that the jitter buffer settings in your example are not in effect for the first group, but would be for the rest. All other settings would be the same for all ports. On Wed, Mar 18, 2015 at 7:20 AM, Andrew Colin and...@convergedgroup.net wrote: 4 Port PRI sangoma a104 *From:* jg [mailto:webaccounts...@jgoettgens.de] *Sent:* Wednesday, March 18, 2015 2:09 PM *To:* Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] 4 Port PRI I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes ;echocancelwhenbridged=no ;rxgain=0 ;txgain=0 callerid=asreceived musiconhold=default group=1 overlapdial=yes signalling=pri_cpe context=extensions channel = 1-15,17-31 jbenable= yes jbforce= yes jbmaxsize= 120 jbimpl= fixed jbresyncthreshold= 1000 PRI or BRI? Which card are you using? Typically the installation script or procedure lets you configure each span. You seem to have 4 spans for either 8 or 128 (EuroISDN) channels. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no
On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome jl...@me.com wrote: Well, it breaks audio for all NAT endpoints, how can I fix this? Local (packet to packet) bridging should not do that. Remote (direct media) can do that. Can you confirm - by looking at a verbose level 4 log - how Asterisk is bridging the two channels? -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk switching bridge to native_rtp even with direct_media=no
Well, it breaks audio for all NAT endpoints, how can I fix this? On 18 Mar 2015, at 15:48, Matthew Jordan mjor...@digium.com wrote: On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome jl...@me.com mailto:jl...@me.com wrote: Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table “direct_media” sets to “no” on all endpoints but it doesn’t help. if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it’s not a right way to fix this issues. A native_rtp bridge is used for more than direct media. It is also used for local native bridging, that is, when you have two RTP capable channels in a bridge and Asterisk does not require the media to flow through its core. The bridge then just performs a packet to packet swap between the two RTP capable channels. Note that on verbosity 4, Asterisk will tell you if the bridge is locally or remotely bridging the two channels. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://digium.com/ http://asterisk.org 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
Re: [asterisk-users] Asterisk 13.2.0 Video issues
Attached is my safe_asterisk script, it is moving the core to some dumpdrop
directory that does not seem to exist.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Wednesday, March 18, 2015 1:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
If you take a look at the safe_asterisk shell script, usually located at
/usr/sbin/safe_asterisk (for CentOS at least), you'll be able to find where
the core files are located. If it's not located there, then you'll need to
look at the Asterisk init script for the scripts location. I hope this
helps.
Regards;
John
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, March 17, 2015 11:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
I see that my asterisk is started with the -g option, the core file I
cannot find on my system (find / -name core*)
I would suspect one of the following:
(1) Asterisk is not actually crashing.
(2) Something is deleting the core files.
(3) The core files are hiding really, really well.
Either way, if you can't get a backtrace, there isn't much we can do to help
with that problem.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
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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
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#!/bin/sh
ASTETCDIR=/etc/asterisk
ASTSBINDIR=/usr/sbin
ASTVARRUNDIR=/var/run/asterisk
ASTVARLOGDIR=/var/log/asterisk
CLIARGS=$*# Grab any args passed to safe_asterisk
TTY=9 # TTY (if you want one) for Asterisk to run on
CONSOLE=yes # Whether or not you want a console
#NOTIFY=root@localhost # Who to notify about crashes
#EXEC=/path/to/somescript # Run this command if Asterisk crashes
#LOGFILE=${ASTVARLOGDIR}/safe_asterisk.log# Where to place the normal
logfile (disabled if blank)
#SYSLOG=local0 # Which syslog facility to use (disabled if
blank)
MACHINE=`hostname` # To specify which machine has crashed when
getting the mail
DUMPDROP=${DUMPDROP:-/tmp}
RUNDIR=${RUNDIR:-/tmp}
SLEEPSECS=4
ASTPIDFILE=${ASTVARRUNDIR}/asterisk.pid
# comment this line out to have this script _not_ kill all mpg123 processes when
# asterisk exits
KILLALLMPG123=1
# run asterisk with this priority
PRIORITY=0
# set system filemax on supported OSes if this variable is set
# SYSMAXFILES=262144
# Asterisk allows full permissions by default, so set a umask, if you want
# restricted permissions.
#UMASK=022
# set max files open with ulimit. On linux systems, this will be automatically
# set to the system's maximum files open devided by two, if not set here.
# MAXFILES=32768
message() {
if test -n $TTY test $TTY != no; then
echo $1 /dev/${TTY}
fi
if test -n $SYSLOG; then
logger -p ${SYSLOG}.warn -t safe_asterisk[$$] $1
fi
if test -n $LOGFILE; then
echo safe_asterisk[$$]: $1 $LOGFILE
fi
}
# Check if Asterisk is already running. If it is, then bug out, because
# starting safe_asterisk when Asterisk is running is very bad.
VERSION=`${ASTSBINDIR}/asterisk -nrx 'core show version' 2/dev/null`
if test `echo $VERSION | cut -c 1-8` = Asterisk; then
message Asterisk is already running. $0 will exit now.
exit 1
fi
# since we're going to change priority and open files limits, we need to be
# root. if running asterisk as other users, pass that to asterisk on the command
# line.
# if we're not root, fall back to standard everything.
if test `id -u` != 0; then
echo Oops. I'm not root. Falling back to standard prio and file max.
2
echo This is NOT suitable for large systems. 2
PRIORITY=0
message safe_asterisk was started by `id -n` (uid `id -u`).
else
Re: [asterisk-users] PRI Callerid Passthrough
My, how embarrassing. I of course meant that as a personal message to Don. But if anyone else knows the answer, I'm interested! lol Cheers, j On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote: Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is free), and I do a TBCT to an international destination, and they stay on the line for ten minutes. Who gets the bill? Cheers, j On 03/18/2015 09:19 AM, d...@donkelly.biz wrote: This depends on what you mean by “not involving the service provider.” If you are literally forwarding calls that come in on the PRI back out on the PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out in the wiki. You need to make sure your carrier supports the feature. When you want to do a “transfer,” you have an incoming call alerting or answered, you initiate an outgoing call (using the originating ANI). You initiate the TBCT and the CARRIER completes the transfer, disconnecting both of your B channels. The carrier will later notify you when the transferred call is done, but I don’t think Asterisk handles this directly. Note that at least one of the calls must be answered when you initiate the transfer. If you are doing “unattended” transfer, you will, typically, leave the incoming call alerting until the outbound call answers, then complete the transfer. An “attended” transfer would generally answer the incoming call, play a message, do some IVR doodling, or chat with an agent then initiate the transfer. Have fun --Don *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan H Qureshi *Sent:* Wednesday, March 18, 2015 7:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] PRI Callerid Passthrough Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough. On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles asterisk_l...@earthshod.co.uk mailto:asterisk_l...@earthshod.co.uk wrote: On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks It depends who your service provider is! Any PRI card can send the commands down the D-channel to set any caller ID you like, but it's still up to the telco whether or not they will honour your request. I know the hard way that BT will only let you identify with a number you're entitled to use. Also, remember if you have a call coming in on a PRI line and going out on another PRI line, that's eating two of your thirty lines . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi http://linkedin.com/in/rhqureshi -- _ -- 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] PRI Callerid Passthrough
Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is free), and I do a TBCT to an international destination, and they stay on the line for ten minutes. Who gets the bill? Cheers, j On 03/18/2015 09:19 AM, d...@donkelly.biz wrote: This depends on what you mean by “not involving the service provider.” If you are literally forwarding calls that come in on the PRI back out on the PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out in the wiki. You need to make sure your carrier supports the feature. When you want to do a “transfer,” you have an incoming call alerting or answered, you initiate an outgoing call (using the originating ANI). You initiate the TBCT and the CARRIER completes the transfer, disconnecting both of your B channels. The carrier will later notify you when the transferred call is done, but I don’t think Asterisk handles this directly. Note that at least one of the calls must be answered when you initiate the transfer. If you are doing “unattended” transfer, you will, typically, leave the incoming call alerting until the outbound call answers, then complete the transfer. An “attended” transfer would generally answer the incoming call, play a message, do some IVR doodling, or chat with an agent then initiate the transfer. Have fun --Don *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan H Qureshi *Sent:* Wednesday, March 18, 2015 7:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] PRI Callerid Passthrough Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough. On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles asterisk_l...@earthshod.co.uk mailto:asterisk_l...@earthshod.co.uk wrote: On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks It depends who your service provider is! Any PRI card can send the commands down the D-channel to set any caller ID you like, but it's still up to the telco whether or not they will honour your request. I know the hard way that BT will only let you identify with a number you're entitled to use. Also, remember if you have a call coming in on a PRI line and going out on another PRI line, that's eating two of your thirty lines . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi http://linkedin.com/in/rhqureshi -- _ -- 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] PRI Callerid Passthrough
This depends on what you mean by “not involving the service provider.” If you are literally forwarding calls that come in on the PRI back out on the PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out in the wiki. You need to make sure your carrier supports the feature. When you want to do a “transfer,” you have an incoming call alerting or answered, you initiate an outgoing call (using the originating ANI). You initiate the TBCT and the CARRIER completes the transfer, disconnecting both of your B channels. The carrier will later notify you when the transferred call is done, but I don’t think Asterisk handles this directly. Note that at least one of the calls must be answered when you initiate the transfer. If you are doing “unattended” transfer, you will, typically, leave the incoming call alerting until the outbound call answers, then complete the transfer. An “attended” transfer would generally answer the incoming call, play a message, do some IVR doodling, or chat with an agent then initiate the transfer. Have fun --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan H Qureshi Sent: Wednesday, March 18, 2015 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Callerid Passthrough Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough. On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks It depends who your service provider is! Any PRI card can send the commands down the D-channel to set any caller ID you like, but it's still up to the telco whether or not they will honour your request. I know the hard way that BT will only let you identify with a number you're entitled to use. Also, remember if you have a call coming in on a PRI line and going out on another PRI line, that's eating two of your thirty lines . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] PRI Callerid Passthrough
The way it's expected to work: Inbound call to our toll-free number, we pay for the call TO US until terminated Inbound call to our local number, caller pays for the call TO US until terminated (if long distance charges apply for caller) In either case, we pay for the outbound call if long distance charges apply As a practical matter, some carriers don't have this figured out very well, so anything could happen! --Don On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote: Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
Hi list , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TLS not working in 11.16
Kindly guide with debugging TLS issue in asterisk 11.16. Compiled from source and works all ok ! Added the below to sip.conf tlsenable=yes tlsbindaddr=0.0.0.0:5061 However asterisk doesn't even listen to port 5061 sudo netstat -anp Kindly guide Thanks Best, Chirag A. -- _ -- 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] res_xmpp.c:3468 xmpp_client_reconnect:
2015-03-18 10:52 GMT-06:00 ricky gutierrez xserverli...@gmail.com: Hi list , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss Hi , I'm trying to apply this patch from the source asterisk asterisk-11.16.0 and when I apply it shows me this message asterisk-11.16.0]#patch -p0 refs patch: Only garbage was found in the patch input. is the correct way to apply the patch or am I doing wrong? regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
2015-03-18 11:13 GMT-06:00 ricky gutierrez xserverli...@gmail.com: Hi , I'm trying to apply this patch from the source asterisk asterisk-11.16.0 and when I apply it shows me this message asterisk-11.16.0]#patch -p0 refs patch: Only garbage was found in the patch input. is the correct way to apply the patch or am I doing wrong? regardss I'm confused this is not a patch, it's just garbage ;), I'm making a connection xmpp with asterisk and not connected, at the cli shows me the message every second: RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client ' RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client ' RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client ' [2015-03-18 12:53:49] ERROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to I hope not bother to write directly matt regardss -- _ -- 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] how asterisk detects silence?
Hello! As I see there is dsp_drop_silence switch in confbridge. Could you tell me how asterisk detects silence? Is it possible to change silence level, so, let's say some not loud enough background noises will be recognized as silence and only loud enough human voice will be recognized as sound? Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?
Hello. Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve. Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11. In asterisk 13 I did not find a handler after the call, but before finalizing the CDR. I tried to call the AGI and there to update the CDR record by unique identifiers. But faced with the fact that there are no needed record in the table yet. To write the data into a separate table and join them may be an option. But do not want to resort to such a decision How do you solve this problem? Dmitriy Serov. -- _ -- 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] Use dialplan variables from MySQL database and replace with value
Jonas Kellens wrote:
Hello
i have the following field (text string) in a MySQL database :
${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}
I read this string form the database and want to have the dialplan
variables to be replaced with the correct content.
Sounds like you need the EVAL dialplan function[1].
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_EVAL
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Dialog-Info Event Support
Amber and Sarosh wrote: Hi I am in need of information about how to configure the sip.conf and extensions.conf for subscribers to support the dialog-info event package rfc 4235. I am using Asterisk 11.7.0.4 version. Also please inform if the phone must have the support for this too? There is no explicit configuration required except ensuring hints are available for the extensions you wish to subscribe to. The subscribing device does need to support the standard as well. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.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] Asterisk only registering at one provider
Dennis Guse wrote: Hey, I am running default Asterisk 11.16.0 on a FreeBSD-Machine. I need to register to several other SIP-Services (actually 3): short sip.conf register = XX@a register = XX@b register = XX@c If I remember correctly this worked quite well, but I now checked the system again and it is only obeying the first register statement. sip show registry only reports the first entry and if I reorder them, this effect stays the same. Did something changed recently in the parsing code for sip.conf or so? Nope, and I'd expect we'd be seeing many bug reports if something like this was occurring. I just did this in my general section in 11: register = meh@tacos register = hola@bob register = yolo@dave And confirmed they appeared as expected: Hostdnsmgr Username Refresh StateReg.Time dave:5060 N yolo 120 Request Sent bob:5060N hola 120 Request Sent tacos:5060 N meh120 Request Sent Does anything show up on the console when chan_sip is loaded? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.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
