> >> If you can provide details, even vague ones, about how you did it, I
> >> can update the WMM package.
> >
> > See http://asterisk.gnat.com/meetme.tgz
> >
> > That's a gzipped tar of our working directory plus the relevant parts of
> > extensions.conf. I xxx'ed out phone numbers and Google
> If you can provide details, even vague ones, about how you did it, I
> can update the WMM package.
See http://asterisk.gnat.com/meetme.tgz
That's a gzipped tar of our working directory plus the relevant parts of
extensions.conf. I xxx'ed out phone numbers and Google interface data.
This
> I have a very old server that is used only for conferences on
> Meetme. To manage the conference rooms we use Web Meetme. Now it is
> time to upgrade everything but since Meetme is no longer available I
> need to find a replacement GUI to manage the conference rooms. Anyone
> know a
We're experimenting with using Asterisk (14.6.0) for video conferences.
This test has three endpoints, a Polycom Trio with its video accessory,
and two desktops running Linphone. The video is all H.264. We're using
Opus for audio on the Linphone Windows desktops and have tried both
G.722 and
I've had two Asterisk crashes today that seem to be caused by errors
where chan->tech_pvt is pointing to something that can't be deallocated
and I think I see a reference count bug in the above function.
It contains:
if (data->chan_old_vsrc) {
> There are certain versions of the Linux kernel that have no support
> under the older version of ESXI. We started having issues under our
> ESXI v4 setup with RH Enterprise and vmware's response was, "It's
> not supported"
"not supported" and "does not work" are not the same thing. ESXI
> The version is licensed and the customer does not want to invest on new
> hardware/software at the moment. If the ESXI version is too old I need
> to give them definitive proof that the segfaults are caused by that but
> since the old elastix has been running there for years they do not
> It was only when I ran AsteriskLint over my dialplan that I noticed this:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET
>
> Hmmm, they both seem to do the same thing. Or don't they?
In some
> Use menuselect's command line (--enable and --disable).
Great idea! How would you recommend generating the set of --enable and
--disable options that differ from the default from a build that was done?
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> Of course, you might run into problems if the later release introduces new
> options (or deprecates old ones) which then aren't going to be in your
> makeopts file
That's my question: how do I reflect the changes that I made to the
defaults in a way that's not dependent on the exact set of
I'd like to be able to save the choices made in menuselect in a way
that they can be tracked in a CM system and applied to a later release
of Asterisk using an automated tool like Ansible. What's the best
way to do that?
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I had three crashes this morning on a divide-by-zero, for example at
abstract_jb.c:1008 in 14.3.0.
Does this ring any bell to anybody?
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Check out the
> The feed function in slinfactory explicitly does not allow frames
> without a data payload to be added to the queue. It would have prevented
> this crash.
Ah, so the fix should really be there, righty?
> I think the underlying issue is that the data pointer is not NULL when
> it sanely should
> All patches need to go into JIRA with a license agreement to be
> accepted.
Understood, but I was using it as an illustration. Note, however, that,
from a legal perspective, a patch such as this has no protectable IP (you
can't copyright the only way of doing something) and the GNU projects
Another crash with a packet:
$10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 324, offset = 64,
src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318,
> I would say this is a bug in func_speex and not in codec_siren14. This
> is because the datalen is zero.
Ah! So, like?
*** func_speex.c.orig 2017-02-13 15:00:19.0 -0500
--- func_speex.c2017-04-06 11:16:03.0 -0400
***
*** 185,189
}
!
I'm seeing Asterisk crashes with the following frame at func_speex.c:188:
(gdb) p *frame
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 232, offset = 64,
src = 0x2ac07413e7f8
I recently upgraded to Asterisk 14.3.0. When playing a SIP file to a
G722 SIP channel (via chan_sip), I get a crash with the following
traceback. This is reproducable:
#0 0x0036fdc30265 in raise () from /lib64/libc.so.6
#1 0x0036fdc31d10 in abort () from /lib64/libc.so.6
#2
> I can't speak for the MRCP guys, but from a difference perspective,
> swapping MRCP from Asterisk 13 to Asterisk 14 shouldn't be too
> difficult. Most of the changes between the two shouldn't affect most
> people's use cases, including projects such as MRCP. I'd definitely
> check with their
When I look at the lastest UniMRCP manual, they only mention as high as
Asterisk 13. Does anybody know if I need to do anything to allow it
to work on Asterisk 14 and, if so, what that is?
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At least in version 12.2.0, the code in cdr.c appears to create CDR
records for each pair of users in a conference. This is quadratic
and would seem to be an issue with large conferences.
I got two Asterisk crashes when a lot of people tried to dial into a
conference. They appear quite related
A Siren codec is not currently available and the one for 12 will not
work. I have no timeframe for when this might change.
So the only option is to build one from the Polycom sources? I'm
already doing this for Siren14 (I forget why).
--
Alas, until we get off our butts, yes. Sorry about that.
Really, we're putting as much effort into fixing things and issues
that affect a lot of people. While siren7/siren14/silk are nice, there
aren't as many people using them as other affected things at this
moment.
Is there something
What is the proper version of the Siren7 codec to use for Asterisk 13.5.0?
Since there's nothing later, does the version for 12.0 work?
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I'm planning on upgrading to Asterisk 13.4 soon and am looking for the
corresponding Siren7 codec. Where do I find it?
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This is an interpolated frame from func_jitterbuffer. It's part of
packet loss concealment. What scenario exposed this?
We were testing for clipping by doing Set(VOLUME(RX)=100) but we were
connecting to a ConfBridge that had a jitterbuffer. This occurred when
the phone (SIP) hung up.
--
I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
id =
CALLERID is a read only variable.
That's not correct. I set it all over the place in my dialplan.
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Question: is there some built-in way to know if macro
feature1-ClientA is defined? Something liken
ExecIfMacro(feature1-ClientA)?macro(feature1-ClientA):Goto(...).
A macro is a context, so DIALPLAN_EXISTS should work if you specify an
extension and priority that's in the macro
What are the cons, if any, of enabling a jitterbuffer?Â
Memory and latency.
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I'm interested in finding out what the source ip is of an invite in the
dialplan (Asterisk 11).
${CHANNEL(recvip)}
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I'm having the error as shown belowÂ
Connecting to 'wss://54.xxx.xxx.xxx:8080/ws' SIPml-api.js?svn=224:1
==stack event = starting SIPml-api.js?svn=224:1
__tsip_transport_ws_onerror SIPml-api.js?svn=224:1
__tsip_transport_ws_onclose SIPml-api.js?svn=224:1
==stack event = failed_to_start
Committed the fix for this leak on Asterisk v12 branch in -r413452.
This leak also applied to Asterisk v11.
Thanks.
Is this for both the one in the talking callback or the one in
handle_cli_confbridge_kick or both (the fix is similar in both)?
--
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
I think the bug is in
That is definitely a leak and the fix looks good.
Thanks.
That leak is most likely the one biting you.
It definitely is.
There is another leak in handle_cli_confbridge_kick() if the
participant to kick is not in the conference.
Confirmed. I missed that one in my code reading. I just
It may show up in 'bridge show all' - but I'd actually expect it not
to show up there either.
Actually, it does. I have a screen full of bridges with 0 channels.
I just tried an experiment where all I have is
exten = 329,1,Answer(1000)
same = n,Confbridge(1234)
with absolutely nothing else
Please go ahead and open an issue and attach the refs log and the full DEBUG
log. That will allow us to understand what's occurring here.
I need to wait until I'm sure this isn't something I caused somehow,
so I need to first understand why I'm seeing this and nobody else is.
--
If the reference count on the bridge is off, you should see the conference
bridge 'hanging around' after the last participant has left.
And how would I be sure this is the case? I did core set debug 1 and
didn't see the debug line about destroying the conference, but it doesn't
show up in
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
That can't be done in the 12.2.0
After an upgrade to Asterisk 12, I'm collecting channels. When I enter
and then exit a conference room, I see:
-- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language 'en')
-- Channel CBAnn/207-067f;2 joined 'softmix' base-bridge
5edb1920-3774-4ba3-8c4d-23e8fd04519c
--
The announcer channel joins/leaves the conference as it has sounds
to play. If the channel still hangs around after the conference is
destroyed then there is a problem.
There's a problem. ;-)
But thanks for pointing to how that's supposed to be handled.
--
If the channel still hangs around after the conference is destroyed
then there is a problem.
Am I missing something obvious: I'm looking in the confbridge_exec
function. I see a conference = NULL line, but no attempt to free
that structure, which is what I understand will destroy the playback
What distro are you building on?
CentOS 5.10.
Both have the libraries listed in install_prereq.
Indeed it has all but 2 or 3 of those libraries (none related to uuid), but
after running that script, it was still missing what it needed for uuid.
Unfortunately, there's no upgrade path from
e2fsprogs-devel is the package that provides uuid.h on centos 5
I tried that first and it didn't seem to. I'm pretty sure I needed
uuid-dce-devel.
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When I run ./configure, it aborts with:
checking for uuid_generate_random in -luuid... no
checking for uuid_generate_random in -le2fs-uuid... no
checking for uuid_generate_random... no
configure: error: *** uuid support not found (this typically means the uuid
development package is missing)
I think you need the libuuid and libuuid-devel packages.
yum list available was not showing any such package.
I installed a few other packages, including uuid-dce-devel and one of them
did the trick, but the install-prereq script wasn't good enough.
--
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
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
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
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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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.
--
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
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
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.
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]
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
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
The way you had things configured Asterisk was prioritizing GSM over
ULAW, so until Jitsi started responding it sent GSM.
I thought I might have seen something like that in the packets, but it
didn't look like it showed up in the SDP negotiations, so seemed
peculiar to me. Unclear why this
I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone
on one of my desktop Windows machines. That machine can either be connected
to Asterisk via an VPN connection (with a static IP address) or not (via NAT).
When it's connected via NAT, all is OK.
When it's connected with
We recently set up a SIP trunk between an office in NY running Asterisk and
an office in Paris (running Alcatel). All works fine if a SIP phone on the
NY system talks to the Paris PBX. But if something on DAHDI (a PRI or
MeetMe) talks to the Paris PBX, there's a low-volume crackling. This isn't
cat proc/interrupts?
http://wiki.openvox.cn/index.php/Troubleshooting_of_PRI_cards
I'm sorry that I wasn't clear: the PRI is fine. It's been in use for
years and hasn't caused any problems. What's new is the SIP
connection between the two offices. And another datapoint: the problem
only
I seem to recall seeing somewhere recently where there was a bugfix
for ulaw/alaw conversion which would cause poor audio.
Hmm. You mean:
https://issues.asterisk.org/jira/browse/ASTERISK-1323
That was quite old, but that is what the noise sounds like.
Have you tried updating your Asterisk
I'm trying to add a Talking: field to the AMI ConfbridgeList event so
that my conference room monitoring will work with Confbridge instead of
having to stay with MeetMe and there's something I don't understand.
When app_confbridge.c calls ast_bridge_features_set_talk_detector, it
passes a *copy*
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
{thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=
I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it. We all know about the missing realtime
linkage. That's a major nuisance, but can be worked around.
More serious is that the
The latest version of res_speech_lumenvox.so doesn't seem to work and
nobody seems to know when a version that works will be available. It
looks to me like this is some sort of timeout issue. Does anybody
have a workaround to allow this to be used? (I know about UniMRCP,
but find it quite
I'm migrating from Asterisk 1.6.2 to 10.7.0. In 1.6.2, I made a small
patch to allow specifying an address for RTP media. That worked. In
10.7.0, this appears to be built in with media_address, but it doesn't
work for me.
My Asterisk server has multiple addresses, all global address on two
I'm getting cycles of repeated crashes which occur and then stop occurring.
Looking at the dumps via gdb shows that something peculiar is happening
that looks like memory corruption:
Program terminated with signal 6, Aborted.
#0 0x003686e30285 in raise () from /lib64/libc.so.6
(gdb) up
#1
Who's responsible for it? Lumenvox is the only place that distributes
it, but they can't do anything with it since they get it from Digium.
However, the current version doesn't work with Asterisk 10.7.1 and the
latest version of Lumenvox software (it appears that a timeout is
being set to zero).
You have hardware echo canceling *outside* of your T1 card?
No, on the card.
The DAHDI layer has some buffering that can help with jitter, but the
default buffers can only handle 80ms of jitter. You can increase this by
setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms
You have hardware echo canceling *outside* of your T1 card?
No, on the card.
Then you definitely don't want 'echocancel=no' set, or you'll disable it.
When I thought that it was echo cancellers fighting each other, that's
exactly what I wanted to do.
--
I'm having a wierd clipping issue with one employee who's using a phone
over a satellite Internet. He was sold that system specifically for use
with VoIP. Ping times show average round-trip time as around 700 ms with a
range of 560 to 841, so considerable jitter.
Things work fine when he's
But so long as you were careful not to copy any of the code you are
going to link against into your Source Code (and why would you, if
you were linking against it?), it only *becomes* a derivative work
*after* it has been compiled.
That's not necessarily true because if you have a work that
Can please the Powers that Be reconsider and add this option to sip.conf?
What Powers that Be? This is open-source software! If you need an
option in sip.conf, just add it!
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how can I get the second character/cipher of an extension ?
If I have : exten = 12345,n,NoOP()
How can I get 2 ?
${EXTEN:1:1}
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