I'm confused exactly what's supported with LDAP and Asterisk. What I want
to do is to have SIP peer information read directly (in realtime) from LDAP.
Can this be done? If so, with what Asterisk versions?
--
_
-- Bandwidth and
It is not a matter of preference, it is actually a rule [1]. Top-posting
is also an annoying practice [2] and NOT the general accepted way to reply.
And that's been the case for at least TWO DECADES. I find it amazing that
this is still being argued now.
--
I'm confused about a few things relating to realtime, SIP and config in
general.
As I understand it, with the exception of extensions.conf, I can either
have a config file completely in text or completely in a database. Is
that correct? I can't find documentation for exactly what switch = does
Who are you hiding them from? Anyone with access to the Asterisk server
can already do far more damage than extracting these passwords.
You may (like we do) want to store config files in a version control system
in a common repository. People who have access to that repository don't
necessary
Anyway, the answer is: No, it's mathematically impossible to do
that. Even if the passwords were stored encrypted, Asterisk itself
has to be able to get the plaintext passwords to send to the remote
server; so the code to decrypt them must necessarily be located on
the machine. And the
How does that improve things? The reason that works with Cisco routers
is because the code that reads that special key file and uses it to
decrypt the other files is closed-source; nobody can see how it works.
As another poster said, that's not true for Asterisk. If Asterisk had
such a
Right. But it really won't help much (except complicating things) if the
user has decent access to Asterisk.
Yes, but we're talking about cases where the user *doesn't* have access
to Asterisk. At many locations, including mine, Asterisk runs on a
machine dedicated for that purpose and only
#include the password (a file the line 'secret=') from a local file on
the file system. The user has no access to it, right?
Right, but we're not talking ONE password, but ANY password. Having
dozens of those files, one for each password, gets to be a real pain
really fast. And you STILL want
- The config file reader looks for strings of the form {enc:string}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the master_key file.
This sounds pretty reasonable, except perhaps that you might only want
to convert
I recognize all the options given yet as I explained before they are not
viable. I do not have the resources to pay someone, I do not have the
expertise to fix this issue because according to an asterisk developer
any fix in that area would be deeply architectural in nature... what
other
exten = s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20)
For instance, a landline number in Paris like 01 42 92 81 00 is read
zero-one, forty-two, ninety-two, eighty-one, zero-zero, where I
assume Americans would read all the digits individually (zero, one,
four, two, etc.)
Maybe
No, conference scheduling is not a feature that we have built
directly into ConfBridge, and I'm debating on what it would look
like.
Scheduling isn't built into MeetMe either, but the fact that it
dynamically reads from a database means that you can write external
programs (such as Web-Meetme)
I just updated libpri 1.4 on my system to the latest from that branch and
my QSIG connection to an NEC SV8300 stopped working. The trace showing
the problem is below:
q931.c:5640 q931_connect: Call 7168 enters state 10 (Active). Hold state: Idle
DL-DATA request
Protocol Discriminator: Q.931
Please create a mantis issue describing this problem.
Pardon my ignorance, but what does mantis refer to?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
I recently tried to update my Aastra 57i to version 3.2 and ran into
a problem. It won't properly register and says contact mismatch.
I added sip contact matching: 2 to aastra.cfg, but that didn't help.
When I look at the SIP trace, but I see is the Aastra sending a
REGISTER and Asterisk
Is asterisk replying differently when firmware 3.2 is used ?
No, but the phone cares with 3.2 and not with 2.6.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Asterisk does indeed send an Options before the OK but my 57i doesn't
seem to mind.
That's odd. It does for me.
Perhaps you need to upgrade firmware on the Aastra phone?
The problem occured when I DID upgrade it! Precisely to the one
you mentioned.
Or turning off qualify for this peer
In that case it suggests it is some setting you have applied to the
phones that is causing it. Can you post the local.cfg server.cfg
files from the phone (removing the passwords from there first)?
Sure: local.cfg is checksums, server information, and:
contrast level: 3
ringer volume: 8
In that case it suggests it is some setting you have applied to the
phones that is causing it.
I just called Aastra tech support. I'm always VERY impressed that the
first person who picks up the phone is very technical. He said that they've
had reports of this issue. The problem goes away
I don't believe you really understand what Open Source means...it
does not mean FREE.
Actually, it DOES mean free, especially since Asterisk is under the
GPL. But, as RMS often says, that's 'free' as in 'free speech', not
'free beer'. That problem doesn't exist in French, where there are
two
FreeCNAM.org is providing a free CNAM API for Open Source PBX users.
This API queries a private CNAM database, and returns standard
15-Character CNAM results. Any entry not already in the database will
be queued for investigation, and added to the database as soon as
information is located.
Try them all again. Remember that this is a static database that has to
'research' numbers it has not seen before.
Well, that doesn't make it very interesting: most calls I'd expect to
get won't have been seen by it before.
By now (a few minutes later), the database should have been
Try them all again. Remember that this is a static database that has to
'research' numbers it has not seen before.
What happens when the CNAM is changed? How often does it go back and poll
the database?
--
_
-- Bandwidth
The system uses real Telco CNAM Dips. Any generic names you get are
from the subscriber's carrier itself. We can only provide what we
ourselves get.
There's more than one CNAM database (aren't there seven?). I would have
hoped that a service such as this would look at a bunch of them and
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}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
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!
--
_
-- Bandwidth and Colocation Provided by
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
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
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 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).
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 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
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 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*
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 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
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 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
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 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 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'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 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
It appears that there are no transcoders from g723 to anything else in
Asterisk 10.7.1. Does anybody know how to fix that?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
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.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
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
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.
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
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
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
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.
--
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.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
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
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
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
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.
--
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.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
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
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
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.
--
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
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)?
--
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
I'm interested in finding out what the source ip is of an invite in the
dialplan (Asterisk 11).
${CHANNEL(recvip)}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
What are the cons, if any, of enabling a jitterbuffer?Â
Memory and latency.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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
CALLERID is a read only variable.
That's not correct. I set it all over the place in my dialplan.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
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 =
I'm planning on upgrading to Asterisk 13.4 soon and am looking for the
corresponding Siren7 codec. Where do I find it?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
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?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
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).
--
101 - 200 of 223 matches
Mail list logo
