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El abr 16, 2014, a las 6:45 p.m., Sean Darcy seandar...@gmail.com escribió:
On 04/15/2014 06:52 PM, Kai-Uwe Jensen wrote:
Oops, had it wrong. Here's how it works for me:
[callcentric-template](!)
type=friend
context=from-callcentric
fromdomain=callcentric.com
If you really want to do it:
1) create a wrapper to asterisk -r
2) pipe the welcome message to /dev/null
3) ???
4) profit
you didn't modify Asterisk.
No you didn't, but you may neverthess have created a derived work. There
are two different legal arguments you can make when two pieces
Modifying a program you have legitimately acquired is Fair Dealing.
The Law of the Land gives you the right to do that, even if the
vendor restricts your exercise of that right in practice by
withholding the Source Code.
That is false. Modifying a program is creating a derivative work.
As
way to know when the far end actually
answers. Polarity
reversals could signal when the far end actually answers, but it isn't
normally available or
standardized. Thus, the line is considered answered when dialing is
complete.
Richard
What does violating license of Asterisk means? Does it means I
won't be able to use any commercial modules or asterisk commercially?
I thought it was open and anyone can change the code?
Anyone *can* change the code. But it's licensed software, just like
most other software. The difference
Of course, any good attorney will never commit to anything. They
will never say it is alright to do X, unless X is do nothing
No, but a good attorney can give guidance as to likely expectations. As
you say, nobody can be sure of something even if it's previously been
established law, but a
line name support is fully supported only by Q.SIG since it
actually
defines how to pass the name. Using the display IE is a defacto standard
but
is only really going to work in the network to user direction.
Richard
correct values. Where is my mistake? Has this function being renamed?
This was just fixed yesterday. See
https://issues.asterisk.org/jira/browse/ASTERISK-23250
Richard
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code is sent to the network and
the call
is hung up. With priindication=inband then a busy tone is sent after a
possible
PROGRESS message.
Richard
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New
.
Has this been done? Can anyone point me to some documentation on how others
have done this?
It's always fun to play
Richard Seguin.
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New
: Command
ActionID: your-optional-id
Command: core show channels
Using the AMI CoreShowChannels action will give you more information
as well as not truncating long strings. It is also the recommended way
to get that information.
Richard
I'm running 10.7.1 (yes, I know it's old, but this may be a problem in
later versions too) and had a conference being recorded via:
Set(CONFBRIDGE(bridge,record_conference)=yes)
The bridge started out at 8KHz despite one HD device. But when the
second came in (G.722), it switched to
event PRI_EVENT_HANGUP
[Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] -- Span
4: Channel 0/2 got hangup, cause 47
Richard
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New
as they are not
likely to change
from version to version where CLI commands can and do change.
Richard
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the Compiler Flags -
Development menu in menuselect.
Richard
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separated list of nat options
(nat=force_rport,comedia). Earlier versions do not support comma separated
nat
settings.
The OP did not specify the Asterisk version and may be using a syntax
inappropriate
to the Asterisk version.
Richard
if the callee (or
asterisk) hangs up.
Any idea why I would see this?
exten = h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS})
Thanks,
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,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS})
Thanks,
Richard Seguin
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to generate a diff file of the committed change:
svn diff -c261498 http://svn.asterisk.org/svn/asterisk change.patch
Richard
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with the lowest channel number.
Richard
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but this is not what we want :-(
You need to use the 'I' Dial option to inhibit the connected line date from
the
Dial from overwriting the values setup by your dialplan. See [1].
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
; channel 1 was created and those settings are still in effect when channel
2
; is created.
Richard
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-- SIP/3230-00a6 is ringing
[Oct 23 17:20:37] WARNING[7593][C-00aa]: chan_ooh323.c:1413
ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60
any idea?
The chan_ooh323 channel driver just needs to be updated to ignore that
indication condition.
Richard
OpenSips
Is there anything that I am missing that probably should be implemented?
Thanks,
Richard
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The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under this
scenario. Basically this setup is for people who are traveling, and may be
using a smart phone at an airport (or something similar). The idea is that our
system can be used to reduce toll costs, and provide access
at this link for some hints about what may be going
on:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Richard
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. That is what
the interception macro/routines are for. That is where you use
CONNECTEDLINE to manipulate the connected line party information as it is
passing through Asterisk. You can also block the connected line update
when the call initially connects using the Dial 'I' option.
Richard
to manipulate party id information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Richard
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On Wed, Oct 2, 2013 at 1:41 AM, Nomad Esst noname.e...@yahoo.com wrote:
Hi list
What is the default value for signalling in
/usr/local/etc/asterisk/chan_dahdi.conf file?
You should always be explicit in setting that value.
Richard
the Q.921 peers are out
of sync with each other. When
you reboot your machines, they will be out of sync with each other for
awhile. I cannot think of a
particular sequence of events off-hand where it would happen though.
Richard
It seems that the ISDN switch you are connected to does not respond to the
RESTART message. You should investigate what the chan_dahdi.conf
resetinterval parameter is set to. See chan_dahdi.conf.sample for a
description.
Richard
-Dial+Handlers
Richard
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How does one do this? We have a particular SIP phone that needs a large
jitterbuffer, but all I can see is how to put it on the *read* side of
the channel.
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applications work with the Monitor
application not MixMonitor. If a Monitor is not active on the channel then
the PauseMonitor and UnpauseMonitor applications hangup the channel.
Richard
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extension.
You need to look at the extension patterns in your dialplan to see where
you have ambiguity between extensions. Are you using the '.' wildcard?
Richard
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in sig_analog.c instead.
Richard
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for MWI?
Other than the limited support for sending MWI messages to ISDN phones on
BRI lines there is no other ISDN MWI support.
Richard
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. That warning message
means Asterisk
is falling behind in processing frames on the channel.
Richard
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and _X.
patterns.
Richard
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national/international prefix settings)
89c: internal counter (i.e. 2204 calls so far)
The other fields are pretty much as described by jg.
Richard
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For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free. :)
www.mhspot.com/sts/
(site is down right now)
And that's related to the problem with it: it hasn't been maintained for
quite a while.
--
answered, it's dead.
I think you are required to send a DTMF 1 to the gv call after answering.
Richard
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the call not come
back to the transferrer.
Richard
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implementations send the name in a separate message AFTER sending
the SETUP message. Asterisk usually puts the call into dialplan when it
receives the SETUP message. Waiting allows a subsequent message containing
the name to arrive and be available in the dialplan for subsequent outward
dials.
Richard
names into the DYNAMIC_FEATURES list.
Richard
On Thu, May 2, 2013 at 3:07 PM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
Add MOH_Class onto the example and the idle channel will hear
music on hold
until the playback is complete on the other channel.
Kevin Larsen - Systems
To get debug messages for the loadable Asterisk *.so module.
In your case since manager.c is part of core you will get debug messages
from manager.c and all other files that are part of the core module.
Richard
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ending (0).
Executing last minute cleanups
voxi@vlr-3:/tmp$
Is this a bug or is this my fault?
It is the wrong command.
Richard
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|intense} span x.
The pri intense debug span x command is an alias for
pri set debug 2 span x that didn't get updated when the real
command was changed to pri set debug intense span x.
This will show the help you need:
bas1104*CLI help pri set debug off span
Richard
in TE/Ptmp mode
An extract from chan_dahdi.conf :
[channels]
layer1_presence=ignore
layer2_persistence=keep_up
You do not need to do both methods. One or the other will suffice.
It is better to use the layer1_presence=ignore method.
Richard
handler on
the SIP/foobar/7003 channel.
Richard
[1] https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers
[2] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
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to the status
you are wanting. The child cannot send that variable value back to the
parent channel. Channel variable inheritance only goes one way: from
parent to child.
Richard
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I'm answering my own email here:
There appears to be a disagreement between the encoding given in the
sources for Siren14 that are downloaded from Polycom (and the ITU, both
are the same) and that implemented by codec_siren14.so. The latter
agrees with the actual device.
The disagreement is
.
Looks like CALLEDTON is the right answer, however it needs mangling
(${MATH(${CALLEDTON}40x7,i)}) to get right values.
I created the following issue to get CALLERID(dnid-num-plan) to have the
same value as CALLEDTON.
https://issues.asterisk.org/jira/browse/ASTERISK-21248
Richard
. (Looks like I missed setting that value. :))
I could only find the value being set to the CALLEDTON channel variable.
Richard
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Do you have transcode_via_sln set in asterisk.conf?
No, but as I said in a later email, I found the problem: when computing the
cost of a path, any downconvert has the same cost. So
siren14 - slin - slin32
is the same cost as
siren14 - slin16 - slin32
which is wrong.
I fixed this
:878 load_resource: Module
'libss7.so' could not be loaded.
what is the problem? Can you please help me to solve this problem?
libss7-trunk cannot be used with any released version of Asterisk.
Richard
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There appears to be a disagreement between the encoding given in the
sources for Siren14 that are downloaded from Polycom (and the ITU, both
are the same) and that implemented by codec_siren14.so. The latter
agrees with the actual device.
If I make a .sln32 file and run the encoder from
Sorry for a possible retransmit: the first was sent from an incorrect
email address.
I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
But the transcoding from siren14 to slin32 is via slin. First, it
seems odd that there's no transcoder directly to slin32 since anything
else
I'm connecting a Polycom SoundStation IP 7000 and trying to use siren14.
I downloaded the codecs and now it will properly transcode to connect
to other phones and play any files that are in .wav format. But when it
tries to play any files with .siren14 extensions, I get complete noise
coming out.
any more information.
Richard
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asterisk
, or FollowMe I
option.
Richard
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
[2] https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
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are created
using the cumulative configuration when the
channel = 1-23
line is processed. Anything after that line will not affect
channels 1-23 since they are already created.
Depending on where in the configuration the snippet you posted is
found will determine if it even has any effect.
Richard
What you say...Richard Mudgett (rmudg...@digium.com):
I've always used dahdi-genconf to just create the
dahdi-channels.conf
and since our PRI is fairly simple (just dump all the channels
into
one
group) it works with dialing with dahdi/g1/(number). I'm trying
?
Is there a variable to be set?
Any ideas will be most welcome
The SendText application is only valid during a call. You are trying
to send a message outside of a call. See MessageSend[1] in Asterisk v10
and later.
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+10
create an issue in the issue tracker:
https://issues.asterisk.org/jira
Thanks.
Richard
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I'm now getting these errors:
[Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-ba7 received frame with invalid timing info:
has_timing_info=1, len=0, ts=426891164, src=RTP
[Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-ba7
It appears that there are no transcoders from g723 to anything else in
Asterisk 10.7.1. Does anybody know how to fix that?
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I'm trying to interface Asterisk with an Alcatel PABX and trying to find
a code that works well. It says it doesn't support ulaw, though it
doesn't reject it. It supports G.729, and that works fine, but we'd prefer
not to use compression.
When I use alaw, the path from Asterisk to the Alcatel
Your sounds might be too loud. We use a lot of custom sounds here and when
the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and
clicks.
Sorry I wasn't clear. This is *always*. I hear it over the call when
there's talking and when there's dead silence (e.g., an empty
When I use alaw, the path from Asterisk to the Alcatel is completely
clean, but the other way has a set of clicks that kind of sound like
old-fashioned audio noise.
[snip]
It's been ages since I experienced that but things to check that come to
mind in no particular order are:
- jitterbuffer settings (try on/off)
I added
jbenable=yes
and get lots of:
[Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-6c7 received frame with invalid timing info:
has_timing_info=1, len=0, ts=371371424, src=RTP
[Jan 24 17:53:41] WARNING[12317]:
Check https://issues.asterisk.org/jira/browse/ASTERISK-12042
I did. But that was with an unofficial G.729. This is with the supplied
alaw codec.
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I'm running Asterisk 10.7.1. In the log, I see:
-- Goto (Conferences,70323,1)
-- Auto fallthrough,
But there is an 'i' extension:
dialplan show i@Conferences
[ Context 'Conferences' created by 'pbx_config' ]
'_[ti]' =1. GotoIf($[${SET(REC=$[${REC}--1])}3]?999) [pbx_config]
://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Richard
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I think the below fixes what I reported earlier. Does that seem right?
*** pbx.c.old 2013-01-23 21:08:51.0 -0500
--- pbx.c 2013-01-23 21:09:31.0 -0500
*** static enum ast_pbx_result __ast_pbx_run
*** 5160,5163
--- 5160,5165
int
+ dst_exten[0] = '\0';
Is this 'construct' prefered over
dst_exten[0] = 0;
or
*dst_exten = 0;
and why?
I'm somewhat of a C pedant here. dst_exten is declared as an array,
not a pointer. So if I want to clear the first byte of the
I'm starting to think about migrating from an old Asterisk box to a
new one and want to use the Asterisk 11 long term support release,
but need Lumenvox integration and I don't see the Asterisk 11
connector bridge for Lumenvox available yet. Lumenvox tech support
says this is under Digiums
I'm the opposite. I'm likely not to scroll down 10 pages to see
the comments at the end.
Wouldn't need to if people trimmed their posts properly.
Precisely (e.g., see above)! Indeed, my sense is that top-posting
*discourages* properly trimming email and that's my main reason against it.
In this properly trimmed example, there's no record of who said what.
When it's relevant, I trim in such a way that that information is
preserved. But I would *never* leave in a header, just the identification
of the person who typed that part. Most mailers, when you include text
from another
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if the new material is on the top or
bottom, but people who top-post and don't trim create really hard-to-follow
emails.
Not really true often times when people do the right thing
I like the example of using that to add somebody to the conference, but
what I don't see is how the dialplan can know what conference the menu
item was called from. I was hoping that some variable might have been set,
but don't see it in the sources. Is the idea to do that outside of the
call to
I'm trying to convert from MeetMe to Confbridge and one part of that is
handling the ending of a conference. So I'm taking the suggestion of
originating a call to the conference and doing:
same = n,Playback(conf-will-end-indigits/${WTIME}minutes)
That crashes Asterisk (with no core dump!) in
I realize the benefits of bottom-posting, especially when posting
inline. But top-posting keeps things in reverse chronological order
so any reader could catch up quickly on any missed messages in the
chain. A new reader scrolls to the bottom and reads up.
What's there to catch up with if you
for to associate a
DAHDI channel is:
Event: DAHDIChannel
Privilege: call,all
Channel: DAHDI/i4/317XXX
Uniqueid: 1356099673.5
DAHDISpan: 4
DAHDIChannel: 3
Richard
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ChanIsAvail is a dialplan application not a CLI command. It also
will not work for what you want in this case.
I'm clearly missing something?
Quite possibly. :)
Richard
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. It also
will not work for what you want in this case.
I'm clearly missing something? Quite possibly. :)
Richard OK - so what I am trying to do is through the AMI interface
ask if channel DAHDI/1 is busy, on hook or available.
How do I tell that
In the past I simply did a core show
is going to affect performance to be a problem
depends on how often you execute the command.
You can also use the AMI DAHDIShowChannels action. If the channel has
a B channel it will be listed with which B channel it currently is
attached.
Richard
-work, s, 4) exited non-zero
on 'SIP/x.x.x.x-0061'
Really could do with a second opinion on this issue as it would quite
a
serious bug if it is one...
The incoming call leg does not appear to be answered yet so I would not
expect the caller to be able to hear MOH.
Richard
will need to monitor AMI for MusicOnHold start/stop events. Dialplan
normally executes before a call is connected/bridged. AGI is really an
external form of dialplan execution.
Richard
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a DAHDI channel with an Asterisk
channel so AMI applications can passively determine the B channel currently
in use. Calls with no-media as the DAHDIChannel do not have an associated
B channel. No-media calls are either on hold or call-waiting.
Richard
. For outgoing calls, the
layers are brought back up as well before the outgoing call is given
to the telco.
Richard
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sometimes can not make outgoing calls. anyone can clarify that?
Use of Asterisk v1.8.7 with libpri is *not* recommended. That
version of Asterisk has a regression in its ./configure script
that does not setup Asterisk to use libpri correctly. Also,
Asterisk v1.8.7 is quite old now.
Richard
?
(dtmf, polarity start, dtmfcidlevel=???)
Richard
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.
Now they are. It is the utilities that have the dependency on
DAHDI. It makes more sense to build from the ground up anyway.
dahdi-linux \__ Hardware level drivers and utilities
dahdi-tools /
libpri - Layer 2/Layer 3 protocols
asterisk
Richard
What's the configuration like for Jitsi in sip.conf?
Just fullname and md5secret plus a phones section that reads:
[phones](!)
type=friend
host=dynamic
context=SIP_Phones
cc_agent_policy=generic
cc_monitor_policy=generic
disallow=all
allow=gsm
allow=ulaw
allow=g729
allow=h264
What version of
What NAT settings are globally in use?
nat=yes
Do you have directmedia turned off or on?
I've tried both ways, but I normally have it off.
This really does indeed feel like a weird NAT issue that is probably
configuration related (probably both in Jitsi and Asterisk).
Except that:
(1)
Yeah this is so weird that packet captures are really needed. A working
call and a non-working call, along with what IP ranges are what.
There are *tremendous* numbers of RTP packets, of course. Are those
captures really going to be useful? That's the problem. If they
*are* going to be
Not that many RTP packets are required. It's just important to see the
SIP signaling and where traffic is coming/going from with the network
topology in mind. That way a clearer picture of where it's saying media
should go to, where it's sending media from, etc can be gleamed. Once
that
Not that many RTP packets are required. It's just important to see the
SIP signaling and where traffic is coming/going from with the network
topology in mind. That way a clearer picture of where it's saying media
should go to, where it's sending media from, etc can be gleamed. Once
that
1. Remove allow=gsm from your sip.conf and reload
That did it! Thanks!
But why should that have been an issue?
2. Disable ZRTP in Jitsi by going into Options - Accounts - Selecting
account - Edit - Security - Uncheck Enable support to encrypt calls.
That was one of the first things I
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