://issues.asterisk.org/jira/browse/PRI-145
Richard
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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
are able to get executed.
The new pre-dial and hangup handler features in Asterisk 11 would
be a solution to your problem. Otherwise, I don't really see a
solution without rethinking your post call processing.
Richard
this?
You could use DTMF blind transfer to transfer the call back into
the dialplan.
Richard
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of this issue, but it appears to be isolated to only this machine.
You might have an A-law/u-law mismatch in the audio path. That
kind of mismatch sounds like static on the line.
Richard
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is assigned to a B channel or a call
moves to a different B channel.
There is also the CHANNEL() dialplan function:
CHANNEL(dahdi_channel)
CHANNEL(dahdi_span)
CHANNEL(dahdi_type)
The DAHDIChannel event and CHANNEL() function are mentioned in the
UPGRADE.txt file.
Richard
,Hangup()
It looks like you are seeing this issue that was fixed earlier
this month:
https://issues.asterisk.org/jira/browse/ASTERISK-20455
Richard
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the value to 17.
Richard
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.
Richard
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CoreShowChannels
Richard
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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
or reserve the emergency priority
line for outgoing only.
Richard
[emergency-services]
exten =911,1,Goto(dialpsap,1)
exten =9911,1,Goto(dialpsap,1)
exten =999,1,Goto(dialpsap,1)
exten =112,1,Goto(dialpsap,1)
exten =dialpsap,1,Verbose(1,Call initiated to PSAP!)
same =n(dialit),Dial(${LOCAL
the code attempts to correct the problem.
Richard
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it looks like the code attempts to correct the problem.
Never mind. I was looking at code that already had the patch for
ASTERISK-20337 included.
Richard
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New
call.
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repeatedly, Asterisk may not have the
correct permissions to access the database.
/var/lib/asterisk/astdb
Richard
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I'm trying to add a Talking: field to the AMI ConfbridgeList event so
that my conference room monitoring will work with Confbridge instead of
having to stay with MeetMe and there's something I don't understand.
When app_confbridge.c calls ast_bridge_features_set_talk_detector, it
passes a *copy*
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
{thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=
the
perfect
solution, however I was unable to play the sound file then connect
the
caller. I would like to bypass the need to press the 1 to accept the
call.
Have you tried the Dial application M or U options?
Richard
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
pbx_extension_helper:
Launching
'Answer'
You appear to be suffering form
https://issues.asterisk.org/jira/browse/ASTERISK-19610
It is fixed in the just released v1.8.17.0-rc1.
Richard
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= _X.,2,Answer
exten = _X.,3,Playback(some-message)
exten = _X.,4,Hangup
I believe that is what the
asyncagi break
command is for.
Richard
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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).
/ is empty. I am
wondering if any good way that I could have some sample
configurations.
Run
make samples
That will copy all of the sample config files from ./configs
into /etc/asterisk with appropriate removal of the .sample
from the filenames.
Richard
,
local_ssrc,
remote_ssrc,
It also will accept nothing or all.
Richard
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systems. The CLI command is not extensible without breaking
existing systems.
Richard
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The AMI action CoreShowChannels deprecated the CLI concise command
because the output of the AMI action is extensible without breaking
existing systems. The CLI command is not extensible without breaking
existing systems. Richard,
Thanks - I tried the CoreShowChannels AMI and it says
notloggedin to something else. There might be a name conflict
when using the DUNDi switch. Otherwise, it looks like a bug.
Richard
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(status)},${FAXOPT(statusstr)})
Any ideas suggestions on how to over come this. Would be appreciated.
Can you quote it like this:
DO-STORE()=${CALLERID(number)},${CALLERID(name)},...
Richard
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.
Richard
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messages.
Richard
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has been in Asterisk since v1.8:
https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29
Richard
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am I missing for it for accept my dialplan remote-id name and
number?
[1] https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Richard
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protocol involved,
connected line updates look like call transfers to the peer.
Richard
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the link did not return.
Richard
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with which each line is put in
energy save mode : a line remainded up during 20s or so and then put
down for a while.
The line will come back up on demand in less than a second.
Richard
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an unplugged line and layer 1 being down is
that they come back up on demand.
Richard
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Both mISDN and DAHDI have drivers for your BRI card. Only one of them
should be loaded. Since you are using DAHDI and not mISDN, you should
load the DAHDI version.
Richard
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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
It seems like much of the problem with virtualizing Asterisk is getting it to
interface with DAHDI cards. Is that correct? As I'm not planning on using
such cards (I'm only interfacing with our broadband connection), would this
make virtualizing more feasible?
Richard
-Original Message
Richard Hiers
Director of IT Services
Covenant Theological Seminary
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virtual configuration
(mem, cpu, etc.)? Any other considerations?
Thanks,
Richard Hiers
Director of IT Services
Covenant Theological Seminary
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. Most supplementary service features and a hangup fix
supported by that version of libpri do not get enabled.
Richard
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chan_dahdi.conf.sample for a description of the new option.
I added a comment to https://issues.asterisk.org/jira/browse/ASTERISK-13176
saying which SVN revisions the patches were committed.
Richard
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parameter.
From chan_dahdi.conf.sample:
; bri_cpe:BRI PTP signalling, CPE side
; bri_net:BRI PTP signalling, Network side
; bri_cpe_ptmp: BRI PTMP signalling, CPE side
; bri_net_ptmp: BRI PTMP signalling, Network side
Richard
. It is otherwise benign if the line is really PTP.
If you can make calls, please create a JIRA issue on the PRI project so the
message level can be reduced. Please attach an intense pri debug output
showing the received MDL messages.
pri set debug 2 span 4
https://issues.asterisk.org/jira
Richard
, there is no obvious dialplan execution
when the calls are redirected, diverted or masqueraded, so we cannot
update the CONNECTEDLINE() information trivially. Or am I missing an
obvious trick?
This is the purpose of the interception macros.
Richard
to start in is configured in
chan_dahdi.conf. See the chan_dahdi.conf.samples file for more
information.
Richard
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it needs to have a built in wait(5) after sending the request
before returning to accommodate switches like yours that need time to
process the request.
Please file a bug report on this so it does not get lost.
https://issues.asterisk.org/jira
Thanks.
Richard
which module is registering the dialplan
items. You can see the combined dialplan with the CLI dialplan show command.
Richard
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On Fri, Mar 16, 2012 at 11:43 AM, Richard Mudgett
rmudg...@digium.com wrote:
snip
pri intense debug:
TEI: 0 State 7(Multi-frame established)
V(A)=31, V(S)=31, V(R)=42
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
T200_id=0, N200=3, T203_id=8192
[ 00 01 54 3e
= INCOMING_CT_STATE_POST_CONNECTED_LINE;
/* Send our subaddress back if we have one. */
if (call-local_id.subaddress.valid) {
Richard
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.
If it is the ROSE_QSIG_CallTransferComplete section then I need to see
more messages to know what is happening in the call. I also do not need to
see intense output because your issue is in the Q.931 level and not the Q.921
level.
Richard
of Asterisk. A lot of parking issues have
been fixed since that version.
Richard
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by making outbound
calls pick channels from the opposite end of the channel range as the
network. This is usually by using uppercase G in the Dial(DAHDI/G1/number)
to the first available channel from the upper end of the channel range.
Richard
; Required for Embarq / CenturyTel
pridialplan
?
Yes. It does look like the carrier is not sending the Reverse Charging
Indication ie.
Richard
Att,
Rafael Saraiva
2012/2/17 Richard Mudgett rmudg...@digium.com
I had not noticed that you switched to direct email earlier.
- Original Message -
Switchtype
as a parking
extension, the first priority of the extension must be the park
application. If the park application is not the first priority of
the extension, then the transfer is treated as a normal transfer.
Richard
. If the park attempt fails, the interrupted bridge will be
resumed.
Richard
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or the default. If the selected
parkinglot does not exist it will then be dynamically created.
See the CLI core show application Park documentation.
4) You are likely running into
https://issues.asterisk.org/jira/browse/ASTERISK-19322
Richard
(reversecharge) value is set from the Reverse Charging Indication
ie received in the incoming SETUP message. Please capture the incoming SETUP
from libpri for the collect call.
pri set debug on span x
Richard
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Reversecharge not appear in debug .
I'm in Brazil , the signaling is different here ?
Please capture the incoming SETUP from libpri for the collect call.
pri set debug on span x
Richard
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How to block collect calls on ISDN trunk?
You need Asterisk v1.8 or later and check the value of CHANNEL(reversecharge)
in your dialplan.
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL
Richard
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= n,GotoIf($[${CHANNEL(reversecharge)} = -1]?allow:block)
same = n(allow),Dial()
same = n(block),Hangup()
Please note that CHANNEL(reversecharge) is only valid on ISDN channels.
Richard
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to the source file.
Richard
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: Call
failed to go through, reason (0) Call Failure (not BUSY, and not
NO_ANSWER, maybe Circuit busy or down?)
kindly help me is it possible to send SMS on Signaling Channel and
how? thanks.
I do not think that Asterisk supports SMS with SS7.
Richard
Hello List,
I have one TOR3-E (E1 version) card from Govarion that i used some years
ago, but it seems company stopped work.
Since website is down.
Is there somebody with good heart that could help me to get a driver for an
x86 and x86_64 for
this card?
Thank you so much,
Richard Palmeron
time.
Richard
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on 'DAHDI/5-1'
-- Hungup 'DAHDI/5-1'
chan_dahdi.conf is mostly just the default with just the lines
defined, nothing too fancy, and this doesn't happen for SIP clients
or remote
phones via the FXO ports.
Any ideas?
You have to hang up the phone too.
Richard
Where to find meaning of /n in Local/6613@from-queue/n ?
See https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers
Richard
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using 'w' to force sending the 'sending complete' IE in an ISDN
setup message.
But I don't know the usage of multiple 'w' in the dialstring.
See https://issues.asterisk.org/jira/browse/ASTERISK-19176
Richard
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static
only when someone is speaking.
Richard
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this is the beep sound. It is likely that the channel was
hung up when Asterisk was trying to play the sound file.
Richard
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on span x for the SETUP message is
needed to figure out why chan_dahdi/libpri thinks that
channel 0 is requested.
Richard
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OK, read all about the patch, thanks for the fix Richard.
I would like to apply this patch to my current 1.8.7.1 but I am
afraid I don't have a clue how.
https://issues.asterisk.org/jira/browse/ASTERISK-17557
Get the patch by following the reviewboard link in the issue and
download
means and is there a way to prevent it
from happening.
No changes has been made on this asterisk box in years (running old
1.4.25 if it ain't broken version)
You have pri intense debug span x enabled.
Disable with pri no debug span x.
Richard
. Just not sure what
I am missing.
Thanks in advance for any help.
This may work now after I fixed this issue last week on SVN v1.8:
https://issues.asterisk.org/jira/browse/ASTERISK-17557
Do you get Extension '%s@%s' doesn't exist\n error messages?
Richard
are already bridged. ECT is automatically initiated
if the chan_dahdi.conf transfer=yes option is set and a call is natively
bridged on the same span.
ECT is also used to update connected line information.
Richard
' setting must also be enabled to allow sending
; the transfer to the ISDN switch, since it sent in a FACILITY
; message.
; NOTE: This should be disabled for NT PTMP mode. Phones cannot
; have tromboned calls pushed down to them.
;
transfer=yes
Richard
-vmail,u)
exten = s,n,Hangup()
exten = s,n(busy),VoiceMail(${MACRO_EXTEN}@officea-vmail,b)
exten = s,n,Hangup()
exten = 1234,1,MeetMe(1234,i)
*Features.conf has this:*
[parkinglot_main]
context = officeA
parkext = 799
parkpos = 800-850
findslot = next
Thanks
Richard Zulu
Twitter
www.twitter.com
. Otherwise, the quotes
are included as part of the name. Everything between the '='
and closing ')' less leading and trailing spaces is part of the
name. For SIP this may be detrimental to the message format
unless the quotes get escaped.
Richard
-invent the background stuff for it.
Asterisk v1.8 added the connected line support not Asterisk v10.
Richard
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/wiki/display/AST/Manipulating+Party+ID+Information
This applies to v1.8 and later.
Richard
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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
(RDNIS).
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Richard
Hi Richard,
1. Could you elaborate a bit ?
Do you imply that the lines bellow were present (or missing) because
I
did somewhere set CALLERID(RDNIS) and that I should use them
, you will need Asterisk v1.6.2 or later. You would also need
libpri 1.4.12 to do this with ETSI(EuroISDN). You would then use
the DAHDISendCallreroutingFacility application *before* you answer
the call to forward/deflect the incoming call back to the network.
Richard
That's
but are just not as efficient.
Richard
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and
libpri v1.4.12. Prior to Asterisk v1.8.x you only have CALLERID(RDNIS).
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Richard
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for manipulation of redirecting number is available
with the REDIRECTING dialplan function in Asterisk v1.8.x and
libpri v1.4.12. Prior to Asterisk v1.8.x you only have
CALLERID(RDNIS).
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Richard
Hi Richard,
1. Could you
commit log -r336658.
Richard
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. Since many SIP and ISDN phones
cannot send DTMF before a call is connected, you need to answer the call
leg to those phones before using Dial with these options for them to have
any effect before the dialed party answers.
Richard
. Timeslot 16
is usually used for signaling. channels = 1-15,17
Richard
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it could be a clocking issue between the two cards then.
Everything needs to use the same clock source.
Richard
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