Hi Doug,
Thanks for the reply. Unfortunately I can't get my telco do do anything
because I can't provide proof that is a problem with their lines.
2011/10/22 Doug Lytle supp...@drdos.info
Richard Reina wrote:
I have a server that is hooked to a channel bank (Adit 600). It has eight
lines
changed at all in the * servers configuration. The server has remained
untouched and has not as much as been rebooted in a couple years.
Has anyone ever heard of a similar problem and can anyone suggest how I
might fix or diagnose it?
Thank you very much for your time,
Richard
it is
already configured to be Pseudo.
Never mind. The chan_dahdi warnings about the Pseudo channel was already
fixed in -r331955 of the v1.8 SVN branch and is in v1.8.7.
Richard
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With the above chan_dahdi.conf and indicate that it is generating these
warnings when Asterisk loads:
[Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure
channel -2 with signaling Unknown signalling -1 ignored because it is
already configured to be Pseudo.
Richard
committed to
v1.8 branch with -r339719. It is fixed in v1.8.8-rc2.
You could simply use the ./configure script from v1.8.6.
Richard
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JT
Sent: Wednesday, October
It looks like you are attempting to manually configure the pseudo channel
multiple times in chan_dahdi.conf. You do not need to explicitly configure
the pseudo channel. The pseudo channel is always created and has no
settable configuration parameters as far as I know.
Richard
say more without digging further into the code.
Richard
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On Thu, Oct 6, 2011 at 11:25 AM, Kyle Sexton k...@mocker.org wrote:
I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP
signaling. Has anyone had any experience with these devices? The
feature cards that Cisco sells can be a little confusing. I'm
thinking something like
to better follow the spec and should be keeping active
calls up.
Richard
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ActionID.
snip
Is there anyway for me to make asynchronous AGIs work? I've tried
searching online to no avail.
The most definitive documentation is always the source code. :)
For AGI this is in the res/res_agi.c file.
Richard
end-point although the call remains up.
I assume this is some sort of firewall/nat/routing issue. Could someone
explain what is possibly going on and perhaps offer a solution?
Cheers,
Richard.
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-routeable
address of the softphone when I turn on rtp debuging.
How can I configure the rtp stream to be sent to the public facing address
of the softphone?
Cheers,
Richard.
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(opvxa1200) for which source code is available.
I think the new behavior is a bug. It is most likely in
chan_dahdi.c:dahdi_request() when it finds that the requested channel
or no channels in the group are available.
Richard
code says Unallocated (unassigned) number. You are dialing an
invalid number. Is the 9 supposed to be in your called number?
Richard
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Hallo Barry,
extensions_additional.conf is supposed to be edited by FreePBX.
Gopal, on using extensions_custom, the SIP phones work however the details
are not captured in the reporting mechanism of FreePBX, which is what I need
most.
Richard Zulu
Twitter
www.twitter.com/richardzulu
http
already laid out dialplan?
Thanks
Richard Zulu
Twitter
www.twitter.com/richardzulu
http://www.linkedin.com/in/richardzulu
Skype: zulu.richard
*
*
*There is no place like 127.0.0.1*
On Thu, Aug 11, 2011 at 4:12 AM, neo haux neo.h...@gmx.com wrote:
Hi
I want to change my old answering phone
Hey,
I have been using asterisk on slackware and had thus come up with my own
dialplan.
I would like to import my dialplan into freepbx+asterisk since I am
switching to that...how can I create my own custom dialplan in freepbx?
Thanks
Richard Zulu
Twitter
www.twitter.com/richardzulu
Skype
/system.conf
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
Richard
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ip 77514810
sp 7fffcd48 error 4 in libc-2.11.1.so[77492000+16b000]
I have used gdb so that I can perform a backtrace however the program
executes and exits without a stack thus not helpful.
Any help is appreciated!
Richard Zulu
Twitter
www.twitter.com/richardzulu
Skype
by: 808blogger
Review: https://reviewboard.asterisk.org/r/1337/
Also -r330505 to fix a ref leak with the above patch.
Richard
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.
There is no signaling to indicate the call is not going to proceed
any further.
Richard
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to the peer channel and opens
the audio path. It is the caller who must recognize any audio message
that their call has been dropped. As far as ISDN is concerned, the
call has not been answered yet so Asterisk must keep waiting.
Richard
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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there.
Richard
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asterisk-users mailing
for
EuroISDN.
It is implemented in EuroISDN(ETSI).
You need Asterisk 1.8 and libpri 1.4.12 to take advantage of the feature.
Richard
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New to Asterisk
In article 296076780.5348.1310743930593.JavaMail.root@zimbra,
Richard Mudgett rmudg...@digium.com wrote:
I would suggest Two B-Channel Transfer (TBCT), transferring to a
unique
number (received as DNIS by the other server) that would identify
the
call
as transferred from
need to capture debug output of earlier events to figure this out.
Richard
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: /* presentation_restricted_null */
ASN1_CALL(pos, asn1_enc_null(pos, end,
ASN1_CLASS_CONTEXT_SPECIFIC | 7));
Richard
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Richard
2011/7/1 Richard Mudgett rmudg...@digium.com
I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i
can't show the callerid name in the way Asterisk == Siemens. I
realized that Asterisk send calleridname in format
namePresentationAllowedSimple to Siemens e
in extensions.conf.
Richard
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asterisk-users
to login
in, it says checking permissions on gui folder and loops. Haven't found much
help on other mailing lists, any direction given in welcome.
Thanks
Richard Zulu
Twitter
www.twitter.com/richardzulu
Skype: zulu.richard
*
*
*There is no place like 127.0.0.1
UTC
ISDN lines connected via Digium TE412P card.
I have faxdetect = both in chan_dahdi.conf in the general
section as well as specifically for the configured spans.
Show us your chan_dahdi.conf
Richard
is practically guaranteed
to cause link issues.
Also make sure that clocking is only supplied by one side of the link.
(The Asterisk1 clock should be slaved to Asterisk2 which should be in
slaved to the Patton link.)
Richard
enabled.
Richard
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asterisk-users
://issues.asterisk.org/jira/browse/ASTERISK-17264
Richard
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http
for SIP
channels.
Richard
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asterisk
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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of these methods should work after doing a quick look a the code.
Does the outgoing call SETUP indicate digital capability?
both show transfercapability DIGITAL
Could be a problem in the media stream handling not being setup for digital
mode.
Richard
.
For completeness, the bug report should have attached:
1) chan_dahdi.conf (and any files it includes)
2) Debug capture files of pri set debug on span x output of a call
attempt for the incoming call leg and the outgoing call leg.
Richard
this can
be interigated?
Check the ChangeLog of your release to see if the fix to add
CHANNEL(dahdi_channel) is present. The fix also added a new
AMI DAHDIChannel event.
Richard
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a the code.
Does the outgoing call SETUP indicate digital capability?
Richard
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block in chan_dahdi.c since I could not find
it elsewhere when studying the code.
Richard
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way traffic?
2) Does the D channel bounce up and down or is it continuously down?
3) Does chan_dahdi complain of CRC errors?
Richard
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there may be other B channels
available for the call.
In old zap school you can do that but in dahdi I don't think you can.
Until unless you create g1 g2 ... Group in chan_dahdi.cfg and map
channels there.
You should be creating groups for your ISDN spans.
Richard
than v1.8.
Richard
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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?
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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
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
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.
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in
addition
to libpri. Is that right?
Yes chan_dahdi.c/sig_pri.c and libpri need to be modified to add a config
option.
Please create a mantis issue describing this problem.
Thanks
Richard
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Please create a mantis issue describing this problem.
Pardon my ignorance, but what does mantis refer to?
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Please create a mantis issue describing this problem.
Pardon my ignorance, but what does mantis refer to?
Mantis is the issue tracker at:
https://issues.asterisk.org
Richard
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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
Now imagine that 1.4 stays at only security level. For first case we
have 2 options: upgrading for security reasons to last version but
then no more voicemail, or staying with 1.4.26. In the second case,
upgrading both servers to test with 1.8. If it's still not working, it
was time
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)
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
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
- 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
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
:21 PM, Richard Zheng rzh...@gmail.com wrote:
Hi,
In ACD queue, is it possible for the agent to take some actions when the
caller hangs up? For example, to let the agent to enter some numbers for
accounting purpose.
Thanks
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
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
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 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?
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What is the proper way to format a caller-ID here in the U.S.?
Is it:
+15705551212
That's the correct one.
I've always seen it +15705551212, but as I understand it the country
code for the US is 011, which to me would indicate you put
011-570-555-1212 as the callback number.
The country
It's kind of low for me. How does one control that volume?
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how can I go from *100* to 100 ?
I know I can do something like ${EXTEN:1} but that way I only get rid of just
one *.
${EXTEN:1:-1} removes the first and last characters of ${EXTEN}.
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I dial 12345678, but only '16 'is received by the asterisk.
You may want to try
relaxdtmf=yes
in chan_dahdi.conf. That fixed a similar problem for me.
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I tried relaxdtmf = yes but has not worked.
If I type very slowly digits are recognized normally.
Then indeed it won't make a difference. If that were your problem, it
likely wouldn't work at any speed.
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This suddenly started appearing and I'm not sure why. Any ideas?
asterisk*CLI module load chan_skype.so
Unable to load module chan_skype.so
Command 'module load chan_skype.so' failed.
[Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error
loading module 'chan_skype.so':
as possible.
Are there well known gotchas that I shoud be aware of?
Thanks in advance,
Richard Stuppi
rich...@stuppi.com
626-221-8010
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New
I'm having a wierd problem. Somewhere around 1-2% of the time, the
first DTMF digit dialed gets dropped. This is occurring during a
SpeechBackground application call. If the caller reenters the digits
when given a second chance, all is OK.
Any suggestions how to debug this intermittent
Is the message played very long/short? I play a lot of my speechbackground
messages with beep in front (speechbackground(beepfoo)) so my user doesn't
start hitting DTMF until the message starts playing.
It's about six seconds. I've seen the problem myself and I'm dialing the
first DTMF digit
1. Vestec, Lumenvox or other?
Vestec
2. How many digits of DTMF are you aiming for (using SPEECH_DTMF_MAXLEN?)
6
3. Are you presenting DTMF back (verbose ${SPEECH_TEXT(0)}) ?
Similar. There's a NoOp that display what was originally that value in
the log.
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Who is the carrier that the calls are flowing in from?
It's a Paetec PRI into an NEC SV8300, then QSIG from there to Asterisk.
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So it's a PRI/DAHDI connection.
Yeah, but with switchtype=qsig, though that difference isn't likely
relevant here.
Is SpeechBackground the first item in the context?
No. There are plenty of others, starting with an Answer(200). Then
a whole bunch of Speech* applications to load grammar
Just for grins, do this command
/bin/grep num sent /var/log/VestecASRE/Port-10500_2010-09-07.log
This should show you all of the DTMF processed by the grammars today.
It doesn't show any. Isn't DTMF processed by Asterisk and not the ASR?
Anyway, I can now reproduce this in a simpler case:
if you use SpeechBackground, DTMF is under ASR control (returned in
SPEECH_TEXT(0) ).
It is returned in SPEED_TEXT(0), but it's still being done by Asterisk,
not the ASR engine.
Anyway, your other test indicates that the DTMF press
used to stop the prompt is being eaten by the ASR or
Hallo Keane,
I truly have a nagios server, up and running 24/7
--
Richard Zulu
Managing Director
Time Information Company
P.O Box 31842
Clock Tower
Kampala, Uganda
www.time.co.ug
Mobile :+256752624006
Skype: zulu.richard
cli commands can help
show channels, show uptime and show sysinfo
here is an example
asterisk -x core show sysinfo
On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu richard.z...@time.co.ugwrote:
Hey guys,
I have my asterisk box running without a gui. I now need to monitor usage,
calls
Hey guys,
I have my asterisk box running without a gui. I now need to monitor usage,
calls, traffic of voice calls on this asterisk server. I cannot now install
a gui because the configs will be wiped out, how can i go about monitoring
all the above?
--
Richard Zulu
Managing Director
Time
At what stage will there be versions of the G.729 codec, res_cepstal,
skypeforasteric, Vestec, etc that'll work with 1.8? It would be good if
people using that software could participate in the Beta.
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WARNING[28505] loader.c: Error loading module 'app_stack.so':
/usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister
This is the gosub issue. It's in app_stack.
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It appears that there's no way to get the return value from a GOSUB into
an AGI script. Is that correct?
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Hi,
i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?
thx
rich
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Danny, Doug
thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
use Read().
rich
On Tue, Jun 29, 2010 at 3:26 PM, Doug Lytle supp...@drdos.info wrote:
Coco Richard wrote:
Hi,
i need to save into a local
Does it exist? Sending email to their support address appears to be a
black hole. They reference a forum, but Google can't find it.
I keep having problems in any grammar than has a an o for zero: it breaks
recognition anywhere NEAR it. For example, if I say two o five, it gets
recognized as
Make sure that you only have the one grammar active when doing your test.
You want the voice engine to basically only have 11 possibilities to chew on
(0-9 plus oh).
I always only load one grammar. In the test I did below, there were
exactly TWO possibilities:
I'm having a lot of problem
We hit this issue and are reviewing the patch to install now...
Any updates?
Nope. I think any of the patches posted to either of the issues will
work, though the official one is obviously the best.
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Maybe your engine is tone deaf. Try showing the ${SPEECH_SCORE(0)} when
you get the foobared result.
I repeated the experiment, this time noting the score, which I output.
This time, the result was always 2 and the score was pretty
high: 711, 743, 752.
--
72 64 20 68
6f 6d 65]
Facility (len=28, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x0B, 0x02,
0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0C, 'Richard home' ]
[1e 02 81 81]
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local
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