Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On Sun, Sep 25, 2011 at 09:39:06PM -0700, Steve Edwards wrote:
On Sun, 25 Sep 2011, Alex Balashov wrote:
Aside from that, is it really that big of a deal? Is it that hard
to learn a new command set and adapt?
Yes, it is.
I confess I'm a 1.2 Luddite so I have close to no experience with
the current CLI. Every time I start using a newer version I get
about 3 or 4 commands into it and then I get sucked into the vortex
of 'core or not core,' what module implements this command?, oh the
hell with it -- I'd be done by now if I used 1.2.
Why should I have to know the name of the module before I can get
help on a command?
Actually knowing the names of modules is easy: Here's what I get after
pressing 'tab' once (Asterisk 1.8)
| sweetmorn*CLI
| !ael agent agiaoccalendar
| cc cdr cel channelcliconfig
| console core dahdi data database devstate
| dialplan dnsmgr dundi event faxfeatures
| file groupgtalk hangup help http
| iax2 indication jabber jingle keys local
| logger manager meetme mfcr2 minivm mixmonitor
| module moh no odbc originate parkedcalls
| phoneprovpri queue realtime reload rtcp
| rtp say sip skinny slasqlite
| ss7 stun timing transcoder udptl ulimit
| unistim voicemail
Now the same without cli_aliases:
| sweetmorn*CLI
| !ael agentagi aoc calendar
| cc cdr cel channel cli config
| console core dahdidata database devstate
| dialplan dnsmgrdundieventfax features
| file group gtalkhttp iax2 indication
| jabber jinglekeys localloggermanager
| meetme mfcr2 minivm mixmonitor modulemoh
| no odbc parkedcalls phoneprovpri queue
| realtime rtcp rtp say sip skinny
| sla sqlitess7 stun timingtranscoder
| udptlulimitunistim voicemail
But anyqway, let's edit /etc/asterisk/cli_aliases.conf and in [general]
set 'template=asterisk12' . Applying the changes ('module reload
res_cli_asiases.so' . Shouldn't it be 'cli reload aliases'?).
Now we get (again, the following are the options suggested by a single
tab completion)
| sweetmorn*CLI
| !add ael agentagi answer
| aoc autoanswer calendar cc cdr cel
| channel clearcli config console convert
| core dahdidata database devstate dialplan
| dial dnsmgr dont dump dundievent
| extensions fax features file flashgroup
| gtalkhangup http iax2 include indication
| jabber jingle keys locallogger manager
| meetme mfcr2minivm mixmonitor module moh
| mute no odbc oss parkedcalls phoneprov
| pri queuerealtime remove rtcp rtp
| save say send set show sip
| skinny sla soft sqlite ss7 stun
| timing transcoder transfer udptlulimit unistim
| unmute voicemail
And under 'show' we have:
| sweetmorn*CLI show
| agentsagi application applications audio
| channeltype channeltypes codec codecsdialplan
| features file function functions globals
| hints image indications manager memory
| profile queue switches translation version
| video voicemail
This means that prototyping any suggested fixes is easy: just provide an
extra template for /etc/asterisk/cli_aliases.conf . No need to dwell
into the C source and touch most of the source files. This only allows
you to alias commands and not rename or change them, but it's a start.
So, if anybody has any actual constructive suggestions, please post
them. Let's not waste time with pointless flame wars.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Sipgate trunk doesn't bridge with other trunk, but works with local extensions
On 02/10/11 21:58, dotnetdub wrote: On 2 October 2011 21:36, Sebastian Arcus s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: Just a follow up. I've opened up udp ports 1-2 on the Linux box (where Asterisk is) and now I have sound. However, bear in mind that the Netgear router/modem which is connected to the Internet (the Linux/Asterisk box is behind it, in a private lan) does *not* have any forwarded ports whatsoever, or DMZ setup. So it is a mystery to me how the Linux box needs ports open, while the Netgear router doesn't. If you look at /etc/asterisk/rtp.conf that bit should make sense. Your netgear probably has a SIP ALG and is looking at the SDP in the SIP messages and dynamically opening / forwarding ports. Considering that in this case my Asterisk is acting as a SIP client/agent connected to sipgate.co.uk - is the sipgate.co.uk end the one which is deciding which rtp ports to use - or my end? Will it make any difference if a change my rtp.conf? Sebastian -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
On Sat, Sep 24, 2011 at 9:35 PM, Bruce B bruceb...@gmail.com wrote: Hi everyone, I don't mean to be rude but honestly which genius comes up with changing the Word to the wise -- if one starts a sentence with I don't mean to be...X your true intentions are to be just that. If you find yourself doing that, please stop. Rethink what you are writing and word it in a more polite manner. You will ruffle less feathers and have a much more constructive dialog. -M -- _ -- 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] Keeping Voice Call Active During Data Connectivity Loss
Greetings- I'm working on a unique Asterisk installation where I've been given a requirement of keeping a voice call active, even during a data connectivity loss. So, let's assume I have remote users connecting to an Asterisk server via sometimes unreliable connectivity such as satellite, wireless, or shudder dial-up. It is certainly possibly this connectivity will go down for a period of time anywhere from a few seconds to a few minutes (or more). During this outage, if a call was already in session, is there any way to prevent the call from be hung up, and simply kept alive until media can begin flowing again? In this situation, both sides of the link would be running Asterisk, 1.4.x or 1.8.x. Is this as simple as telling both sides not to hangup at a lack of media? Are the steps the same whether using SIP or IAX (preferred IAX in this usage case, unless SIP is specifically required)? Thanks! --Tim -- _ -- 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] Keeping Voice Call Active During Data Connectivity Loss
Assuming that you don't have some sort of reconnect protocol going on like SIP headers, a native-bridge to a local channel might do the trick for you. If you are using DAHDI, you might be out of luck. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Monday, October 03, 2011 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Keeping Voice Call Active During Data Connectivity Loss Greetings- I'm working on a unique Asterisk installation where I've been given a requirement of keeping a voice call active, even during a data connectivity loss. So, let's assume I have remote users connecting to an Asterisk server via sometimes unreliable connectivity such as satellite, wireless, or shudder dial-up. It is certainly possibly this connectivity will go down for a period of time anywhere from a few seconds to a few minutes (or more). During this outage, if a call was already in session, is there any way to prevent the call from be hung up, and simply kept alive until media can begin flowing again? In this situation, both sides of the link would be running Asterisk, 1.4.x or 1.8.x. Is this as simple as telling both sides not to hangup at a lack of media? Are the steps the same whether using SIP or IAX (preferred IAX in this usage case, unless SIP is specifically required)? Thanks! --Tim -- _ -- 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] Keeping Voice Call Active During Data Connectivity Loss
Maybe you could use a very simple sollution like a meetme room - you have only to be creative with the dialplan. Ioan www.modulo.ro -- _ -- 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_1.8.7.0-1digium1 100% CPU
On 11-10-02 07:53 AM, Sassy Natan wrote: Hi Group I have added the following to my /etc/apt/sources.list deb http://packages.asterisk.org/deb natty main deb-src http://packages.asterisk.org/deb natty main deb http://packages.asterisk.org/deb natty-proposed main deb-src http://packages.asterisk.org/deb natty-proposed main When installing asterisk_1.8.7.0-1digium1~natty_amd64 using the following: apt-get install asterisk-config asterisk-core-sounds-en-gsm asterisk-dahdi asterisk-dbg asterisk-dev asterisk-doc asterisk-h323 asterisk-mobile asterisk-mp3 asterisk-mysql asterisk-ooh323 asterisk-voicemail my system get into 100% cpu on of it cores. Once I remove the asterisk-h323 system is working fine! However I need the h323 codec and chan since I'm using it. As you mentioned, this is not specific to the packages but also affects a version of asterisk compiled directly from source. I suggest you open an issue on the tracker. If you can determine when the issue started, it would also help. -- 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
[asterisk-users] Asterisk 1.8 Manager Perl Script Problem
Hi All,
Trying to upgrade some call servers, in the lab making sure all my
applications work, ran into an issue with some manager perl scripts
that pull and reset database info, it seems the command and result
responses have changed but I'm not sure how to resolve. My scripts
are using CPAN Asterisk::Manager; and are working fine on asterisk
1.2.32 but not on Asterisk 1.8.6.0.
Here is the abbreviated script where 1.2.32 is astman1 and 1.8.6.0 is astman2:
#!/usr/bin/perl -w
use strict;
use warnings;
use Getopt::Long;
use Asterisk::Manager;
##setup manager connections##
my $astman1 = new Asterisk::Manager;
$astman1-user('username');
$astman1-secret('password');
$astman1-host('10.10.14.101');
$astman1-connect || die $astman1-error . \n;
my $astman2 = new Asterisk::Manager;
$astman2-user('username');
$astman2-secret('password');
$astman2-host('10.10.14.102');
$astman2-connect || die $astman2-error . \n;
##query databases for cnam count##
$astman1-sendcommand(Action = 'DBGet', Family = 'cnam', Key = 'count');
my @result1 = $astman1-sendcommand(Event = 'DBGetResponse');
my $cnamcount1 = 0$result1[7];
$astman2-sendcommand(Action = 'DBGet', Family = 'cnam', Key = 'count');
my @result2 = $astman2-sendcommand(Event = 'DBGetResponse');
my $cnamcount2 = 0$result2[7];
##total cnam count##
my $cnamtotal = ($cnamcount1+$cnamcount2);
##reset cnam count to 0##
$astman1-sendcommand(Action = 'DBPut', Family = 'cnam', Key =
'count', Val = '0');
my @result101 = $astman1-sendcommand(Event = 'DBGetResponse');
my $cnamreset1 = $result101[1];
$astman2-sendcommand(Action = 'DBPut', Family = 'cnam', Key =
'count', Val = '0');
my @result102 = $astman2-sendcommand(Event = 'DBGetResponse');
my $cnamreset2 = $result102[1];
##disconnect the manager connections##
$astman1-disconnect;
$astman2-disconnect;
print Total CNAM Count for last month is $cnamtotal\n\n;
-end script
The response from the 1.8.6.0 server is Response Success Message
Result will follow but is seems the actual response is not pulled
into $result2. The DBPut command works fine and I get a success
response. I've searched through all the upgrade docs but nothing
mentions command syntax changes.
Any help is appreciated.
Thanks.
JR
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Engineering for the Masses
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[asterisk-users] Delay before ringing from PSTN`s call
Hi I am testing a degium TDP400P (2fxo+2fxs) on my asterisk I configured incoming calls from pstn to ring my SIP phone (extension : 100) cat extensions.conf ... [from-pstn] exten = s,1,Dial(SIP/100,10) same = n,VoiceMail(100,u) root@PC-debian:/etc/asterisk# cat dahdi-channels.conf ... ... ... ;;; line=1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default ... ... ... What I don`t understand is why the SIPphone rings after 3 secondes after Astererisk detects the incoming call. Moreover, after hanging off the external caller the SIPphone continue to ring for 3 seconds. I did those modifications in the file /etc/asterisk/chan_dahdi.conf without improuvement ( After restarting Asterisk) [channels] cidstart=ring immediate=yes faxdetect=no usecallerid=no Here is the debug from Asterisk console *CLI -- Starting simple switch on 'DAHDI/1-1' -- Executing [s@from-pstn:1] Dial(DAHDI/1-1, SIP/100,10) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/100 -- SIP/100-0001 is ringing == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1' -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- _ -- 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
