Displaying 20 results from an estimated 6000 matches similar to: "mysql sip realtime"
2015 Jun 09
2
Manipulate extension state in 1.8.x
Hi
Is there any way to set the presence state of a peer to in-use in asterisk
1.8?
The idea is to integrate DND buttons on phones to BLF.
Regards
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
2008 Mar 25
1
How to obtain SIPCHANINFO variables within custom application?
Hello,
How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough
variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my custom
application, because there's no chan_sip.h and I can't include it into my
application (maybe there's other way?).
I can do like this:
exten =>
2008 Jun 03
8
Any reason to *not* use AEL? (Also, MixMonitor q)
I am building a new Asterisk server here at the office, and I'm
wondering if there are any downsides to creating my dialplan with AEL.
It seems more intuitive (to me), but I'm not sure if there are any
pitfalls I need to be aware of first.
We use this for internal extensions, 8 pots lines, and our answering
service which gets about 500 incoming calls a day down our T1.
Also, one more
2007 Dec 19
3
Realtime logic in Asterisk 1.4.16.1
Hello,
I have configured one provider in Asterisk Realtime DB without username and password, only host=<providers_IP> and ipaddress=<providers_IP>
Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/NUMBER at Provider)
In Asterisk 1.4.15 debug I see that Realtime engine is using query:
[Dec 20 00:02:15] DEBUG[14634]:
2015 Feb 16
1
Asterisk 11.6. SIP realtime lost peers after 'sip reload'
Hi, list.
We have a problem with loss peers after 'sip reload', our configuration:
Asterisk 11.6-cert1, SIP realtime peers, sip.conf:
- rtcachefriends=yes
- rtsavesysname=yes
- rtupdate=yes
- rtautoclear=yes
When we do 'sip reload' , peers are removing from available.
Before `sip reload` :
srv-pbx2*CLI> sip show peers
Name/username Host
2008 Mar 07
3
Silencing VoiceMail() app in * 1.4.10
Hi there,
Googling through the archives it looks like I'm the ferst person to want
this...
My aim is to set up a voicemail application with a custom greeting before
*AND AFTER* the punter has left the message.
Right now the relevant section of my dialplan is like this:
exten => 2,1,Playback(/media/asterisk/answerphone-en)
exten => 2,n,VoiceMail(2000,s)
exten =>
2008 Jun 27
2
How to pass variable between 2 Asterisk servers over IAX2
Hello,
Anybody can advice how to pass variable between 2 Asterisk servers over
IAX2?
With SIP I can use SipAddHeader.
How do to the same with IAX2?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
2015 Jan 08
2
queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
2011 Apr 13
11
Realtime SIP & peer status
Hello,
I'm using SIP realtime with MySQL DB.
Is it possible to get the status of the SIP peer (free / calling) from
this realtime DB ?
If not, is there another way to obtain the call state of a SIP peer ?
Kind regards,
Jonas.
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2010 Jul 06
2
ARA : Realtime or not ?
Hello list,
what is the use of realtime SIP peers when you always need to reload the
sip configuration as if you were just putting your SIP peers in sip.conf ??
My SIP peers are now defined in a mysql-DB and when I add a mailbox in
the field 'mailbox', the change is not active untill a do a "sip reload"
or a "module reload chan_sip.so".
Doing a "sip
2011 Jan 24
1
extconfig, realtime, and SIP
I'm confused about a few things relating to realtime, SIP and config in
general.
As I understand it, with the exception of extensions.conf, I can either
have a config file completely in text or completely in a database. Is
that correct? I can't find documentation for exactly what "switch =>" does
but is that only in the dialplan and a way to have it partly from a file
and
2006 Apr 02
1
morcdr v0.1 released
CDR Stats Analyzer and Report generator
It's a rework of famous Asterisk Stats written by Areski.
The main goal for this project is to concentrate more on PDF reports
(managers love them!).
Later more functions will be added. Please test it and send suggestions how
to improve it.
Licence: GPL
Examples, demo and more info on homepage: http://www.paskambink.lt/mcc
Regards,
2011 Aug 11
5
Trouble with *8 Pickup
We have a client that has sporadic problems with the *8 pickup facility.
The server they are using is 1.8.5 and they are using Snom phones.
Every now and then when they try to do a pickup from another phone they
get a forbidden message on the phone and I can see the following in the
logs.
[Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL
[Aug 8 11:51:53] ERROR[19314] astobj2.c:
2010 Aug 15
6
Realtime Context
Hi,
I'd like to be able to create contexts in real-time when I add new clients to my asterisk box.
Currently, I have to create a blank context in extensions.conf and add:-
switch => Realtime/@
Is there any way to avoid the step of creating the blank context and simply include all the entries from the database?
Thanks
Dan
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2010 Mar 29
5
Continue a dialplan when the client hang up the call
Hi all,
When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately.
At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc.
Thanks in advance,
Giang
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2009 Mar 19
2
Script to softly restart Asterisk each midnight to clean locked channels
As Asterisk has inner problems and channels very often locks we have such
script to restart Asterisk each midnight.
We (our clients) mostly use v1.4.18.1. We can't upgrade to newer versions
because there are too much changes which would brake our system
(realtime/sip/iax2/cdr/etc/etc).
Script soft hangups all alive channels in dirty way then kills Asterisk and
starts it up.
Hope
2014 Jul 21
1
TLS, STRP and ARA
Hi
I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP.
However, we exclusively use the asterisk realtime architecture using the
mysql connector.
Looking at tutorials we have to set encryption=yes and transport=tls for
any peer we want encrypted traffic for.
Having a look at contrib/realtime/mysql/sippeers.sql from the source code
shows that the encryption column is
2011 May 19
3
Manager logged on/off messages
Hi
Is there a way I can stop Manager logged on/off messages from going to
the console/logs without losing all the other information I need?
Regards
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2010 Dec 17
2
Asterisk Freeze In 1.4 realtime
Has anyone seen the following in 1.4 (1.4.17)
We have istances when the number of sip channels in use multiples up
(eg: we have 40 channels in use, and then it will jump to 80, then 100+
and it will keep going upwards) and in doing this, all the channels
which are in use at that time are simply cut off or frozen.
The only way for us to get everything back to normal is via a hard
restart of
2011 Feb 28
2
Asterisk 1.8.3-rc3 and one way audio
I've just installed 1.8.3-rc3 on a test server as we really needed that
deadlock involving REFER fix on our server but now I'm having an odd
issue with one way audio with a specific type of call.
If I do extension to extension calls there is full 2 way audio.
If I route in an incoming call through numbers provided by our SIP
provider there is no inbound audio (mobile to * SIP extension)