Aligning presence over multiple servers is not simple and require some
changes on the dialplan and some custom code to transmit the state from one
server to the other.
The BLF on the phone is displayed using the hint of an extension. To be
able to manually manage the hint of an extension, you
Starting with Asterisk 11 basic AMI commands are documented
(https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation). There are always some
AMI changes as far as actions and events go. A complete documentation doesn't seem to be
possible because there are so many modules and it
I haven't tried it, but the res_corosync module states it will sync device
state across servers.
https://wiki.asterisk.org/wiki/display/AST/Corosync
On Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini ldard...@gmail.com wrote:
Aligning presence over multiple servers is not simple and require
I am trying to setup an extension in asterisk which dials an extension
on the NEC DSX. i.e. If an asterisk user dials 402 I want it to
connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. (
404 would be the NEC DSX sip account that I have the credentials for
).
[402]
It seems very good! I am going to test it when I have a bit of time!
Leandro
2013/11/14 Ryan Wagoner rswago...@gmail.com
I haven't tried it, but the res_corosync module states it will sync device
state across servers.
https://wiki.asterisk.org/wiki/display/AST/Corosync
On Thu, Nov 14,
Hello,
when calling a group of SIP peers like this :
Dial( SIP/inno0SIP/inno4SIP/inno6,30)
is it possible to have a SIP header added for just 1 of these SIP peers,
like only for SIP/inno0 but not for SIP/inno4 and SIP/inno6 ??
I know the function SipAddHeader(), but when I use this in the
Use a LOCAL Channel and redirect that one peer through some dialplan
Something like this:
Dial(LOCAL/inno0@addheaderSIP/inno4SIP/inno6,30)
[addheader]
exten = inno0,1,SipAddHeader(foo)
exten = inno0,n,Dial(SIP/inno0)
exten = inno0,n,Hangup
On Thu, Nov 14, 2013 at 9:35 AM, Jonas Kellens
Stephen More wrote:
I am trying to setup an extension in asterisk which dials an extension
on the NEC DSX. i.e. If an asterisk user dials 402 I want it to
connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. (
404 would be the NEC DSX sip account that I have the credentials for
On 14 November 2013 16:35, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
when calling a group of SIP peers like this :
Dial( SIP/inno0SIP/inno4SIP/inno6,30)
is it possible to have a SIP header added for just 1 of these SIP peers,
like only for SIP/inno0 but not for SIP/inno4 and
Hey all,
I've got a question about including the SIP method MESSAGE in the SIP
Allow header sent by Asterisk (version 1.8). I found a post
(http://forums.asterisk.org/viewtopic.php?f=1t=83638) on this question,
but there was no response. I've also done a bunch of searching for
asterisk sip allow
Hello,
I was trying to use a queue in linear order and to provide the exact order
of members to dial by adjusting the uniqueid value. Obviously it doesn't
work and it seems an old problem:
https://issues.asterisk.org/jira/browse/ASTERISK-18480
Realtime configuration can't identify orders in the
I was in the same place as you are now and following links helped me, thanks to
Matthew Jordan;
*
https://wiki.asterisk.org/wiki/display/AST/New+in+1.8#Newin1.8-AsteriskManagerInterface
*
https://wiki.asterisk.org/wiki/display/AST/New+in+10#Newin10-AsteriskManagerInterface
*
From: Leandro Dardini [mailto:ldard...@gmail.com]
Sent: Thursday, November 14, 2013 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Queue linear unordered feature when using realtime
Hello,
I was trying to use a queue in linear order and to provide
Hi list, I need some help to improve my cdr, now in my company are
asking me how
to know which of my phone numbers are most used when receiving calls from
the PSTN and incoming the IVR
was thinking about using userfield field, and I'm trying to do, I have at
the moment 4 channel DAHDI
; DAHDI
Hi,
On one of our servers, we're having problems with incoming faxes. The
connections come in through a PRI into a Digium TE820 card. 'fax show stats'
yields the following:
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts: 0
Receive
Try this. The warning and notice error's are basically telling you whats wrong
[in]
exten = s,1,Set(CDR(userfield)=23X6)
same = s,n,Goto(in2,s,1)
Mike
From: troxlinux [mailto:xserverli...@gmail.com]
Sent: Thursday, November 14, 2013 2:53 PM
To: Asterisk Users Mailing List - Non-Commercial
thnk , works great ..
2013/11/14 Michael Gilleran mgille...@realtyim.com
*Try this. The warning and notice error’s are basically telling you
whats wrong*
[in]
exten = s,1,Set(CDR(userfield)=23X6)
same = s,n,Goto(in2,s,1)
Mike
*From:* troxlinux
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