[asterisk-users] Software echo with Jack module
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, If I create a jack metro that beeps every second and I connect it to the jack port that goes to the phone headset, I can hear it, as expected, in the phone. However, I can also hear it coming back in the jack port for the phone mic. I've tried different versions, different phones, a softphone... I'm pretty sure that this is not a hardware echo. Has anyone experienced this problem or can give me some advice to try to solve this? If I change the txgain and rxgain, the echo gets better, but it doesn't go away.. Thanks, Murray -- _ -- 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] chan_motif / res_xmpp problems
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 server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the -4 command line option to use IPv4 only just in case Asterisk doesn't like to see IPv6 ICE candidates. I try clicking to make an audio-only call from Jitsi. In the Asterisk logging (xmpp set debug on) I see the incoming session-initiate XML stanza but Asterisk does not send any XML back. I definitely have icesupport=yes in rtp.conf and I've tried it with and without specifying a TURN server from each end. Is this working for anybody? What does your motif.conf configuration file contain? If it is not configured then it will not be associated with the account and the Jingle support will not be present. It is largely based on the default config: [default](!) disallow=all allow=ulaw allow=h264 context=incoming-motif ; Default context that incoming sessions will land in ;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer ;maxpayloads = 30 ; Maximum number of payloads we will offer [asterisk](default) disallow=all allow=alaw allow=ulaw transport=ice-udp connection=asterisk context=incoming_xmpp and in xmpp.conf: [asterisk] type=client serverhost=some-host username=asterisk@some-host secret=-- usetls=yes usesasl=yes status=available statusmessage=I may be available timeout=5 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? Is there any way I can enable ICE debugging? -- _ -- 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] chan_motif / res_xmpp problems
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 work which forwards logging messages from the PJ core into Asterisk log messages. -- 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] chan_motif / res_xmpp problems
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. In 12 there is a module as part of the PJSIP work which forwards logging messages from the PJ core into Asterisk log messages. Has ice-udp been tested against Jitsi already? If not, could you please comment on the clients it has been tested with so I can see if they work against my Asterisk setup? -- _ -- 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] chan_motif / res_xmpp problems
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 within 11. In 12 there is a module as part of the PJSIP work which forwards logging messages from the PJ core into Asterisk log messages. Has ice-udp been tested against Jitsi already? It was tested when the code was originally written, since then I don't know if others have used it. The chan_motif channel driver is primarily used by people for Google Voice. -- 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] Call Identifier Logging
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}});
Dial(Motif/google/+${EXTEN:3}@voice.google.com,,r);
hangup;
}
And worked perfectly.
It would be interesting, the developer team add a variable to channel with
this data.
Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2014-07-21 18:59 GMT-03:00 Steven Wheeler swhee...@usinternet.com:
Hello,
I am working on upgrading from Asterisk 1.8 to Asterisk 11.6. One of the
features we are excited for is Call Identifier Logging
https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging.
However, it doesn't appear that this new Call ID is accessible from the
dial plan. Ideally we would like to store this Call ID in the CDR. Does
anyone know if this is possible?
I could do something like this, but it seems like a terrible hack:
same = n,Set(CALLID=${SHELL(asterisk -rx core show channel ${CHANNEL} |
grep ' Call Identifer' | egrep -o 'C-[0-9a-f]+')})
Also as a side note, in the core show channel output ' Identifier' is
misspelt as ' Identifer'
*Steven Wheeler*
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Re: [asterisk-users] Call Identifier Logging
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
Subject: Re: [asterisk-users] Call Identifier Logging
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}});
Dial(Motif/google/+${EXTEN:3}@voice.google.commailto:exten%3a3...@voice.google.com,,r);
hangup;
}
And worked perfectly.
It would be interesting, the developer team add a variable to channel with this
data.
Att,
Rafael dos Santos Saraiva
[http://www.linkedin.com/img/webpromo/btn_liprofile_blue_80x15_pt_BR.png]http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2014-07-21 18:59 GMT-03:00 Steven Wheeler
swhee...@usinternet.commailto:swhee...@usinternet.com:
Hello,
I am working on upgrading from Asterisk 1.8 to Asterisk 11.6. One of the
features we are excited for is Call Identifier
Logginghttps://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging.
However, it doesn't appear that this new Call ID is accessible from the dial
plan. Ideally we would like to store this Call ID in the CDR. Does anyone know
if this is possible?
I could do something like this, but it seems like a terrible hack:
same = n,Set(CALLID=${SHELL(asterisk -rx core show channel ${CHANNEL} | grep
' Call Identifer' | egrep -o 'C-[0-9a-f]+')})
Also as a side note, in the core show channel output ' Identifier' is misspelt
as ' Identifer'
Steven Wheeler
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Re: [asterisk-users] Call Identifier Logging
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 -- _ -- 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] Call Identifier Logging
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 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Identifier Logging On Tue, Jul 22, 2014 at 12:45 PM, Eric Wieling ewiel...@nyigc.commailto:ewiel...@nyigc.com wrote: Making LinkedID available in the dialplan would also be useful. LinkedID is already available in the dialplan: CHANNEL(linkedid) Richard -- _ -- 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] Call Identifier Logging
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 funcs/func_channel.c
468 else if (!strcasecmp(data, linkedid)) {
469 ast_channel_lock(chan);
470 if (ast_strlen_zero(ast_channel_linkedid(chan))) {
471 /* fall back on the channel's uniqueid if
linkedid is unset */
472 ast_copy_string(buf,
ast_channel_uniqueid(chan), len);
473 }
474 else {
475 ast_copy_string(buf,
ast_channel_linkedid(chan), len);
476 }
477 ast_channel_unlock(chan);
While useful, that doesn't solve the problem of being able to store the
channel's logging identifier in CDR.
Steven Wheeler
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Re: [asterisk-users] Call Identifier Logging
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 Logging
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 funcs/func_channel.c
468 else if (!strcasecmp(data, linkedid)) {
469 ast_channel_lock(chan);
470 if (ast_strlen_zero(ast_channel_linkedid(chan))) {
471 /* fall back on the channel's uniqueid if
linkedid is unset */
472 ast_copy_string(buf,
ast_channel_uniqueid(chan), len);
473 }
474 else {
475 ast_copy_string(buf,
ast_channel_linkedid(chan), len);
476 }
477 ast_channel_unlock(chan);
While useful, that doesn't solve the problem of being able to store the
channel's logging identifier in CDR.
Steven Wheeler
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Re: [asterisk-users] Call Identifier Logging
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 swhee...@usinternet.com:
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 funcs/func_channel.c
468 else if (!strcasecmp(data, linkedid)) {
469 ast_channel_lock(chan);
470 if (ast_strlen_zero(ast_channel_linkedid(chan))) {
471 /* fall back on the channel's uniqueid if
linkedid is unset */
472 ast_copy_string(buf,
ast_channel_uniqueid(chan), len);
473 }
474 else {
475 ast_copy_string(buf,
ast_channel_linkedid(chan), len);
476 }
477 ast_channel_unlock(chan);
While useful, that doesn't solve the problem of being able to store the
channel's logging identifier in CDR.
*Steven Wheeler*
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Re: [asterisk-users] Call Identifier Logging
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 funcs/func_channel.c
468 else if (!strcasecmp(data, linkedid)) {
469 ast_channel_lock(chan);
470 if (ast_strlen_zero(ast_channel_linkedid(chan))) {
471 /* fall back on the channel's uniqueid if
linkedid is unset */
472 ast_copy_string(buf,
ast_channel_uniqueid(chan), len);
473 }
474 else {
475 ast_copy_string(buf,
ast_channel_linkedid(chan), len);
476 }
477 ast_channel_unlock(chan);
While useful, that doesn't solve the problem of being able to store the
channel's logging identifier in CDR.
Steven Wheeler
Where is this documented?
It does not appear to be documented. However, there is a reference in the
Asterisk: The Definitive
Guidehttp://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Monitoring_id246945.html.
Steven Wheeler
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Re: [asterisk-users] Call Identifier Logging
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 to gain access to an internal variable. It is also
likely to create bugs in the future if someone changes the output of that
command. For instance if they fix the typo in Call Identifer.
Steven Wheeler
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Re: [asterisk-users] Call Identifier Logging
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 :)
How about something like:
asterisk -rx core show channel SIP/spa841-0003\
| awk '/Call Identifer/ {gsub(/[][]/,); print $3}'
Of course, a dialplan function would be best.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Call Identifier Logging
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'[][]' '/Call Identifer/ {print $2}'
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Call Identifier Logging
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:
asterisk -rx core show channel SIP/spa841-0003\
| awk -F'[][]' '/Call Identifer/ {print $2}'
This is one of those features that is embarrassingly simple and yet,
unfortunately, was overlooked.
Ideally, it'd be in the CHANNEL function.
If anyone is curious, the accessor function you want is
ast_channel_callid. It returns the callid ref bumped, so you do have
to make sure you decrement the ref count using ast_callid_unref. You
can print the callid to the CHANNEL function's buffer using
ast_callid_strnprint.
--
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Call Identifier Logging
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 asterisk@sedwards.com:
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'[][]' '/Call Identifer/ {print $2}'
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] asterisk performace 64bits
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 performace of the asterisk? tks Eduardo -- _ -- 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 performace 64bits
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 -- _ -- 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
