Hi,
I'm running Asterisk 11.7.0 with app_jack and I'm finding a software echo.
When I answer a call, I get two dahdi jack ports to connect the call. One
for the phone mic and the other one for the phone headset. The audio that I
send to the phone, comes back in the other jack port. For example,
On 21/07/14 15:12, Daniel Pocock wrote:
On 21/07/14 14:33, Joshua Colp wrote:
Daniel Pocock wrote:
I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.
I am trying to call from Jitsi to Asterisk through a Prosody XMPP
Daniel Pocock wrote:
snip
FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1
Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?
Nope.
Is there any way I can enable ICE debugging?
Not within 11. In 12 there is a module as part of the PJSIP
On 22/07/14 18:20, Joshua Colp wrote:
Daniel Pocock wrote:
snip
FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1
Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?
Nope.
Is there any way I can enable ICE debugging?
Not within 11.
Daniel Pocock wrote:
On 22/07/14 18:20, Joshua Colp wrote:
Daniel Pocock wrote:
snip
FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1
Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?
Nope.
Is there any way I can enable ICE debugging?
Not
Hi
I tried this in ael:
_000. = {
Proceeding();
callident = ${SHELL(asterisk -rx core show channel
${CHANNEL} | grep Call Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d]
-f1 | cut -d\n -f1):0:-1};
NoOp(${callident}});
Making LinkedID available in the dialplan would also be useful.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Tuesday, July 22, 2014 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, Jul 22, 2014 at 12:45 PM, Eric Wieling ewiel...@nyigc.com wrote:
Making LinkedID available in the dialplan would also be useful.
LinkedID is already available in the dialplan: CHANNEL(linkedid)
Richard
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Which version was that added? I don’t see it on my 11.10.0
[daffy-01 ~]# asterisk -rx core show function CHANNEL | grep -i link
[daffy-01 ~]#
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Tuesday, July 22, 2014
Making LinkedID available in the dialplan would also be useful.
LinkedID is already available in the dialplan: CHANNEL(linkedid)
Which version was that added? I don’t see it on my 11.10.0
[daffy-01 ~]# asterisk -rx core show function CHANNEL | grep -i link
[daffy-01 ~]#
According to
Where is this documented?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Wheeler
Sent: Tuesday, July 22, 2014 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Identifier
Try this:
CDR(userfield) = ${SHELL(asterisk -rx core show channel ${CHANNEL} |
grep Call Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d] -f1 | cut -d\n
-f1):0:-1};
Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2014-07-22 15:08 GMT-03:00 Steven Wheeler
Making LinkedID available in the dialplan would also be useful.
LinkedID is already available in the dialplan: CHANNEL(linkedid)
Which version was that added? I don’t see it on my 11.10.0
[daffy-01 ~]# asterisk -rx core show function CHANNEL | grep -i link
[daffy-01 ~]#
According to
Try this:
CDR(userfield) = ${SHELL(asterisk -rx core show channel ${CHANNEL} | grep
Call Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d] -f1 | cut -d\n
-f1):0:-1};
Att,
Rafael dos Santos Saraiva
This isn't a suitable long term solution as it requires launching several
external processes just
On Tue, 22 Jul 2014, Steven Wheeler wrote:
Try this:
CDR(userfield) = ${SHELL(asterisk -rx core show channel ${CHANNEL} | grep Call
Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d] -f1 | cut -d\n -f1):0:-1};
Not really interested in this topic, but invoking 6 processes seems a bit
excessive
On Tue, 22 Jul 2014, Steve Edwards wrote:
How about something like:
asterisk -rx core show channel SIP/spa841-0003\
| awk '/Call Identifer/ {gsub(/[][]/,); print $3}'
Or:
asterisk -rx core show channel SIP/spa841-0003\
| awk -F'[][]'
On Tue, Jul 22, 2014 at 2:29 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Tue, 22 Jul 2014, Steve Edwards wrote:
How about something like:
asterisk -rx core show channel SIP/spa841-0003\
| awk '/Call Identifer/ {gsub(/[][]/,); print $3}'
Or:
Really, a dialplan function would be best. I too don't like of an idea of
using a external process to get internal variables, but when necessary...
:(
Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2014-07-22 16:29 GMT-03:00 Steve Edwards
Hello,
I'm running Asterisk on a CentOS 64-bit server. . Asterisk if I compile
using the ./configure --libdir=/usr/lib64 instead of ./configure have a
relative gain performace.? Has anyone done any comparison?
Is there any way in the compilation or even in settings that I can improve
the
If it's a 64-bit CentOS, then you'll have 64-bit binaries by default. Just compare the size of
the binaries with both options. Years ago there could have been occasional problems, if you had
32-bit and 64-bit binaries on your machine.
jg
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