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scripts:
echo /usr/local/lib /etc/ld.so.conf.d/usr_local.conf
/sbin/ldconfig
This worked for me on a fresh CentOS 6.4 installation where I didn't use
the install_prereq script, and thus was having your same issue. Hope this
helps someone in the future!
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as I can tell, no AMI would be needed...
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:5609
messagecount: SQL Execute error!
Could you post a sanitized version of your res_config_mysql.conf and
extconfig.conf files? I'm thinking maybe you've got an error in there
somewhere that's causing this error.
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the error. Otherwise, post whatever
you're new error is.
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=?]
I'm not sure on this. Hopefully someone else can help.
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of the variable
you want that specifically contains the source media IP, but I imagine you
can pull it with the SIP_HEADER function, or possibly the CHANNEL(recvip)
function.
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]: rtp.c:849 process_rfc3389: Comfort noise
support incomplete in Asterisk (RFC 3389). Please turn off on client if
possible. Client IP: 209.220.119.19
Is the other asterisk server under your control?
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*name*.
I usually only see the peer if I make a call to the peer or the peer makes
a call first.
Do you have rtcachefriends=yes in your sip.conf?
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On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.netwrote:
When I compare my total minutes on the bill from VoIP Innovations, to the
number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count
of minutes. I'm wondering why it's there.
Are there different
it
Have you tried maybe setting up the entire call in an AGI that will execute
the desired script as you make the dial command? Or, you could look at
running the M or U options in your Dial() command to execute a macro or
gosub routine when the call is connected?
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-internal.
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to another DID, or to a voicemail box, or to a user-defined
function, etc.
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every Thurs:
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what I'm seeing.
Is anyone else seeing this issue? Should I open an issue on the tracker?
Anyone see something obvious I missed?
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On Wed, Jan 16, 2013 at 1:37 PM, Danny Nicholas da...@debsinc.com wrote:
Same issue exists with 11.2
I've created issue 20945 to track this, at least for 1.8.20.0.
https://issues.asterisk.org/jira/browse/ASTERISK-20945
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on that inside your AGI.
Faxdetect will detect the CNG tone after the call is answered and
automatically route for you. It's not the kind of thing you want to set on
a call by call basis. If you're looking to detect a CNG tone inside your
AGI, I'm not sure what mechanism is available for that.
--Warren
call so I can hangup the call at a later time.
Since you're using AMI to originate the calls, you should then also be able
to add an ActionID to the originate command. You should then be able to
lookup the call in AMI using the ActionID.
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If I remember correctly, dahdi dummy was removed and the functionally added by
default when you load dahdi with no TDM cards installed. I could be wrong
though.
What do you need dummy for?
Thanks,
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On Oct 23, 2012, at 10:28 AM, Jerry Geis ge...@pagestation.com wrote:
I
(from the new version)? I'm talking
examples of the table rows in question. Is it recording the call, just
labeling it answered instead of unanswered? I've never seen asterisk
simply not record a call in whatever CDR backend you're using, regardless
of disposition.
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problems than it solves.
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. It's an average of how long the answered
calls had to wait, not an average of all current calls waiting on hold. At
least, that's my understanding of the issue...
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I've listed above utilize Queuemetrics and they
both love it. The licensing is very reasonable for the market and they offer
free evaluations as well.
Thanks,
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On Sep 27, 2012, at 1:02 PM, Mitch Claborn mitch...@claborn.net wrote:
Warren - that coincides with what I am
upper level support at your telco and ask
them how and when they send your callerid information...
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=
'NO ANSWER' to your query. This is all pretty basic SQL Query writing, not
specific to asterisk...
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)
That is, if you're just looking for numeric callerid. If you also want to
account for extra characters, you can add those to the first part of the
filter.
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in advance.
Is there a specific reason you want to access the realtime data through the
Manager API and not directly from the database itself? It seems like the
Manager API would add an extra layer to whatever you're trying to
accomplish.
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outage that takes out both the
asterisk server and the internet, and your asterisk box comes up but your
internet doesn't, this won't work.
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, from the time you start to the time it all ends?
Is this happening on even just adding one person to the confbridge? Does
it happen when you add more than one person to the bridge? Does everyone
hear the beep, or is it only in the mixed audio?
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http
the asterisk sip
trunk.
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On Aug 9, 2012, at 6:05 PM, Eduardo Giacoman giaco...@gmail.com wrote:
Danny, thanks for your input...
Can you tell me if I am wrong with the following or give me a brief guide of
what to look at?
I was planning on using Asterisk
.
** **
I believe that Switchvox has customized the voicemail email into html.
Has anyone ever tried this? Thanks,
/Josh
What about changing 'mailcmd=' to a shell script that rewrites the email in
the format you want before sending it to sendmail?
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that using custom
contexts are not helping in you situation, perhaps you can expand on what
the actual issue is that you're experiencing, and we can try to help
troubleshoot from there.
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}/${EXTEN})
Swap _XX for whatever your outbound extensions would be...
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other users on the
Flowroute network, I'm not sure. But in general, once your call leaves the
Flowroute network, the only way to get the CNAM info is from a CNAM dip to
the national database (I don't recall the actual name).
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)
[outgoing-dial]
exten = _NXXNXX,1,Dial(SIP/1${EXTEN}@flowroute)
exten = _1NXXNXX,1,Dial(SIP/${EXTEN}@flowroute)
${callidnum} is a variable from my SIP peer (setvar=callidnum=7133437300).
This always passes my proper phone number when I make outbound calls.
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output as a response to this email, and we can
diagnose from there.
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New
custom
context:
include = from-internal
Be sure to do all of this in extensions_custom.conf, that way it doesn't
get overwritten whenever you issue a reload in the GUI.
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having to dial a 1
include = custom-local-only
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On Jul 4, 2012, at 9:20 PM, Thomas Perron thomas.per...@gmail.com wrote:
What am I missing please? sip show registry shows that I am registered.
What are you missing? A question, or at the least, a description of whatever
problem you are having? Also, a meaningful subject that somewhat
attch in gsm format)? if
yes can you show me how please?
I don't think that was an option in 1.2, but I haven't used 1.2 in so long
I may be off. Hopefully one of our resident 1.2 luddite's will see this
and have a more definitive answer for you.
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...
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the calls, that would be very helpful as well.
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On Wed, Jun 20, 2012 at 12:30 PM, Chris Gentle gent...@gmail.com wrote:
On Wed, Jun 20, 2012 at 12:14 PM, Warren Selby wcse...@selbytech.comwrote:
As you said, GV and asterisk integration is unstable at best. I haven't
worked with it in a while, to be honest. But, with all that being said
settings. Instead try using defaultuser.
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(No answer, going to voicemail for 123456789)
exten = 1005,n,Voicemail(123456789@default,u)
exten = 1005,n,Hangup()
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the infrastructure setup as well, that would be
helpful.
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On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com
wrote:
Hmm, I tried calling myself (the asterisk voicemail) from another SIP
provider, same problem. What always works reliable
and will populate
it with the result of the channel variable ${queuenum}, which you should
set before you enter the queue. If you're using MySQL for your CDR
storage, I believe you have to create the column first for the new field.
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.
Almost the exact same issue you're reporting...
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to defirenciate from internal call or external
call?
Just a thought, but maybe set a variable in your sip.conf for each internal
peer, and then check for that variable before you do the SipAddHeader
command (using an ExecIf statement).
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+ and Iaxmodem implementations.
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:
header with that value.
Isn't that likely to cause race issues if for instance he Originates 30
calls all at the same time? I would think a better approach would be to
set a unique channel variable for each originated call and track based on
that?
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Are you able to add a Wait(2) at all to the beginning of your incoming
dialplan? A lot of missing callerID problems are because the callerID
value gets sent after the initial call signaling comes in.
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()
exten = *280,n(disable),Verbose(Disabling night mode)
exten = *280,n,Set(DB(nightmode/enable)=0)
exten = *280,n,Set(DEVICE_STATE(Custom:lamp)=NOT_INUSE)
exten = *280,n,Playback(disabled)
exten = *280,n,Hangup()
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DAHDI,
then run ./configure on Asterisk Source and then install? Or did you
install asterisk first, then DAHDI? I've successfully used DAHDI with
Asterisk 1.4, so there must be some issue. Please give us information
about how you installed everything.
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On Apr 11, 2012, at 5:40 PM, list...@gmail.com wrote:
And your examples should work for 1.8.10 correct?
I just typed those out really quick, so there may be some syntax errors, but
generally yes they should all work with 1.8.x.
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, or simply EXEC a GoTo command (either way
works).
Ultimately, I would go with the AGI option, because that then allows you to
do things like use a database to store your routing information, use case
statements, create routing loops, etc. It's up to you though.
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not registering, I
tried allowing externip as my routers IP, even then its not getting
registered.
What settings are you currently using, and what does your infrastructure look
like?
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. Priority will always be either a number or an 'n'.
exten = EXTENSION,PRIORITY,COMMAND
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with different patterns matching at
the same point in an extension? Where is your priority 1?
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output before the failure? I've seen this type of error
before and a lot of the time it has to do with the insecure= settings
being used.
Which version of asterisk are you using?
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/${EXTEN},30)
exten = _23XX,n,Dial(SIP/2300,30)
This doesn't take things like DIALSTATUS into account, however it
accomplishes the same goal of having a fallback number, if that's what you
want. If you want to add a check for DIALSTATUS, just do it for each
pattern.
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)
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On Fri, Mar 30, 2012 at 8:43 AM, Mikhail Lischuk mlisc...@itx.com.uawrote:
**
Warren Selby wrote 29.03.2012 22:46:
To do this, you change your features.conf setting like so:
parse = *9,peer/both,Macro,Parse
The same result when I changed to Macro. I believe that it's true
= _X.,n,Set(finaldst=${EXTEN})
exten = _X.,n,Goto(mainmenu,s,1)
exten = h,1,Verbose(Hanging up)
exten = h,n,Set(CDR(dst)=${finaldst})
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to chan sip in 1.6.2, I remember that being a selling
point on a 1.6.2 upgrade for a client of mine about a year and a half ago.
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, with the right nat= setting, you
may be able to tcpdump the communication with that peer and get the private
IP address so that you can then attempt narrow it down. This is not a long
term solution, obviously, as it would create a gaping security hole, but
it's worth a shot.
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, if you don't have any TDM interface cards). The
dahdi_dummy virtual device was removed a few versions ago as it was
redundant - just installing DAHDI provided the same timing source that
dahdi_dummy did.
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with us your musiconhold.conf configuration please.
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a few times with Polycom phones and the SipAddHeader()
application in Asterisk. There's plenty of guides out there with details
on how to do this.
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?
Is that the actual localnet= statement you're using, because to my
understanding that is not the proper format to use (should be
localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
y.y.y.y is your subnet for your local network).
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Why not try set a variable under each device in sip.conf to the same as the
endpoint name then Dial(SIP/${CustomVar})?
Thanks,
--Warren Selby, dCAP
On Dec 15, 2011, at 7:03 PM, Barry Miller asterisk-us...@notanet.net wrote:
Hi all,
In sip.conf:
[general]
regcontext = autoreg
,
make install, and make samples. This should add the necessary modules to
asterisk, as well as the sample config files.
This of course assumes you've got mysql and it's development packages
installed.
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.
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On Nov 23, 2011, at 6:11 AM, virendra bhati virbh...@gmail.com wrote:
Hi Gohar,
As per you suggestion I make context into AEL file and working file.
But I do little bit RD on that case I make same context into both
files(.conf and .ael) and asterisk read 1st
7bccfde7714a1ebadf06c5f4cea752c1:VirendraBhati
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that is, and we can compare the differences. My guess is it has something
to do with Playback having an automatic Answer(), and whatever you're
Goto'ing doesn't...
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If you call DumpChan from an AGI you should be able to read the response
programmatically and then dump the data into a database. Cleans up your
dialplan but requires some scripting or programming knowledge (php, perl, bash
or even C) in order to write the AGI.
Thanks,
--Warren Selby, dCAP
.
Thanks,
--Warren Selby, dCAP
On Nov 2, 2011, at 9:09 AM, Christian Tardif christian.tar...@servinfo.ca
wrote:
On 02/11/2011 05:04, Anton Kvashenkin wrote:
Turn off faxdetect on this peer.
2011/11/2 Christian Tardif christian.tar...@servinfo.ca
Hi,
I have a 1.6.2.6 fax installation
Look at upgrading to at least 1.6.2 or 1.8, these both have newer timing
sources that don't rely on dahdi. Also, if conferencing is a big deal, look at
10, this contains a complete rewrite of ConfBridge which doesn't require dahdi
for mixing at all.
Thanks,
--Warren Selby, dCAP
On Nov 1
of 's' in the Voicemail()
application as skip intstructions, not silent. Sorry, just one of
those things that made me go hmmm. :)
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work? In other words, if you changed it
from Queue() to Dial() your sip extension (or whatever means you have of
answering the call), does it work then or does it also hang up after two
seconds?
What version of Asterisk and Zaptel are you using?
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http
think what you're looking for is a GoSub that ends with a Return(value).
You then can pull up the value in ${GOSUB_RETVAL}. But I may be
misunderstanding what you're wanting to do.
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can pull this information from a dialplan native
application, but you could probably write an AGI that pulls this information
for you. The AGI Environment data includes things like the current channel
in use, which should be able to start you off in the right direction.
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be more reasons hidden in your extensions.conf, if you
want to share it maybe someone here can go over it and spot anything that
sticks out?
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the asterisk cli, the following:
module unload app_queue.so
module load app_queue.so
And report back any error messages that may pop up.
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!
Have you tried setting the SIP peer for the 3G connection to only allow
g729?
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Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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it was hooked up to on the card.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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New to Asterisk? Join us for a live
Check for any kind of SIP interference from the end user's router.
Thanks,
--Warren Selby, dCAP
On Oct 14, 2011, at 2:38 PM, Adam Robins arob...@pharmacentra.com wrote:
Thanks I will do that. The user is remote, so I must first RDP into her home
network and do it from her PC.
From
open a ticket on the issue, respond here with the issue id, I'd like to
track it.
Thanks,
--Warren Selby, dCAP
On Oct 11, 2011, at 11:39 PM, Asterisk Man theasterisk...@gmail.com wrote:
Thanks Warren,
I have been using X-lite for member and the system from where it is running
was down
as Status: 5
(Unavailable).
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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New to Asterisk? Join us for a live introductory
://www.virtualbox.org/manual/ch01.html#intro-installing
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Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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and then make a few test calls until you've
got a successful callerID and an unsuccessful callerID, then paste an
example of each call (the complete call, from the initial inbound to the
hangup on the other end) in another email and we can review them for any
discrepancies.
--
Thanks,
--Warren Selby
this
connection into your asterisk system.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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New to Asterisk? Join us for a live
match one of these
patterns, FreePBX is going to look internally for a dial pattern to match
against, and if it doesn't find one there, it will end the call.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
a SIP trunk and an IAX trunk as well.
Thanks,
--Warren Selby, dCAP
On Aug 16, 2011, at 10:16 AM, Morten M. Hansen m...@bellcom.dk wrote:
Hi
I'm hoping someone could comment on how our setup will perform under
larger loads.
Its a quite simple setup, with Asterisk 1.6.2 on Debian 6 on an EC2
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