Hello,
Is there any way to make Asterisk not drop calls that are overwhelming to
the outbound trunk capacity and have it queue that call till the time the
trunk is free to take a new call?
Thanks.
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make gives this:
codec_amr.c: In function 'amrtolin_sample':
codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this
function)
codec_amr.c:227: error: (Each undeclared identifier is reported only once
codec_amr.c:227: error: for each function it appears in.)
codec_amr.c: In
Hello,
I'm trying to fetch outbound SIP PROGRESS Reason cause code in the
dialplan,
Asterisk 1.8.26.1 sip show settings:
Q.850 Reason header:Yes
Store SIP_CAUSE:Yes
However, i'm not getting any value in the dialplan variables, any
successful users of this feature?
--
Hello,
If anyone has successfully compiled asterisk with:
app_rtsp
codec_amr
mp4_play
mp4_save
app_transcode
h324m_call
Please share the versions of OS software, and libraries used.
Lets make this thread useful so that all tried and tested video resources
of asterisk can be found in one place
Hello,
There seems to be a problem with asterisk cdrs when calls are generated via
AMI Originate using Local channels.
Asterisk writes CDR as soon as A party off-hooks. Resulting in very
inaccurate billsec and duration values.
Expected CDR in case of local channel origination should be 2
Use allowguest=no
And define ACLs for every SIP account.
And obviously, fail2ban for blocking suspicious IPs.
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New to Asterisk? Join us for a live
Hello,
Can anyone tell me the format for meetme list concise command, so that I
know what field is what (separated by '!'s)
Thanks
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I don't get what the 'F' option is for. Its not proper to exit a context
and then reenter the conference as admin
Isn't there any other way to do actions such as kick/mute/unmute users by
admin dtmf trigger?
On Fri, Jun 1, 2012 at 3:47 AM, Steve Edwards asterisk@sedwards.comwrote:
On
** **
Dan
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *[Digital^Dude] ®
*Sent:* Thursday, August 15, 2013 4:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users
/log/messages contains only AGI messages.
On Thu, Dec 27, 2012 at 10:48 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Thu, 27 Dec 2012, [Digital^Dude] ® wrote:
I disabled all logger channels but still it logs to /var/log/messages.
Any hints?
What version of Asterisk?
What does
I disabled all logger channels but still it logs to /var/log/messages.
Any hints?
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Hello,
I am using asterisk 1.6.x and 1.8.18.0 (LTS) on a CentOS 5 boxes. I'm using
php AGI for incoming calls and after an hour of running, there is a
mismatch b/w active channels (ss7) and active calls in core show channels
count. Active calls are much higher than even the physical SS7 channel
Hi all,
Can asterisk 1.8.x give me MixMonitor recordings of 16Khz sampling rate?
Any help would be appreciated.
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Hello everyone,
I am trying to apply
thishttp://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diffpatch
on chan_ss7-2.1.0 for RingBack tone but its not accepting and
throwing errors:
Hunk #1 FAILED at 704.
Hunk #2 FAILED at 715.
I have done the patch modifications manually in
Is there a tool integrated with asterisk which can give us the pitch of the
utterance?
On Thu, Jul 12, 2012 at 3:09 PM, Satish Barot satish4aster...@gmail.comwrote:
Hi Akhilesh,
Probably this link would give you some idea on ASR. With the help of it,
add some logic in dialplan to develop an
Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
CentOS release 5.5 (Final)
RAM: 4 GB
CPU: Dual Xeon 2.66 GHz
Asterisk is running as root
data seg size (kbytes, -d) unlimited
file size (blocks, -f) unlimited
pending signals (-i) 38912
max locked
Machine specs: CentOS release 5.5 (Final)
RAM: 4 GB
CPU: Dual Xeon 2.66 GHz
Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
*CLI ulimit core
Core file size (core) is effectively unlimited.
*CLI ulimit data
Program data segment (data) is effectively unlimited.
*CLI ulimit descriptors
Hello,
How can I set a hard limit to the number of Local channels asterisk can
spawn?
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Hello,
Asterisk under 90% load of SS7 calls can only withstand the voice
broadcasting for 30 minutes. After around 30 minutes, it stops receiving
any call hits via AMI. No errors are reported. Giving it a minute's rest
makes it work for another 30 minutes.
Can anyone hint to what may be
Hello all,
I have all configurations done and cdr_odbc works fine.
However, cdr_adaptive_odbc doesn't work and I get the following error:
relocation error: /usr/lib/libmyodbc3-3.51.26.so: symbol strmov, version
libmysqlclient_15 not defined in file libmysqlclient.so.15 with link time
reference
cdr_odbc works fine but I have trouble using cdr_adaptive_odbc.
asterisk fails to start when I uncomment 'first' and 'second' groups. Here
is the error:
WARNING[22722] cdr_adaptive_odbc.c: No such connection 'MySQL1' in the
'first' section of cdr_adaptive_odbc.conf. Check res_odbc.conf.
Hello,
I need to return to the original call leg that I wanted to transfer the
call to. in case the destination IVR has put me in a rather long queue.
Please suggest a way I can hang up the atxfer leg and return to the first
call leg.
The hangup parameter in dial app using '*' key works only
You can unload the features module maybe
On Wed, May 23, 2012 at 9:18 PM, Eduardo Pimenta
e...@akivasoftware.com.brwrote:
Hi Guys,
is there any way to disable all Asterisk Features? We are having false
dtmf detections and randon calls being put on-hold and suspect that dtmf
features is
Hello,
I am able to get the call recording file path of each call in the CDR. How
can I get the realtime call recording streaming?
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Please share if anyone has encountered this cdr issue with call transfer.
On Fri, May 18, 2012 at 5:32 PM, [Digital^Dude] ®
millennium@gmail.comwrote:
Hello,
I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine.
Each CDR entry of calls that are transferred
Hello,
I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine.
Each CDR entry of calls that are transferred is repeated once. Every field
including uniqueid, calldate, billsec, duration, src, dst, channel,
dstchannel is exactly the same.
Besides adding a constraint in the
I am using AMI Originate on asterisk 1.8.11 on SIP channels. I have set
unanswered=yes in cdr.conf because I want to log NO ANSWER and BUSY
calls.
The issue is, that if a SIP peer is not registered, and an originate
request is made for that peer, a null cdr entry is made as follows:
Be it meetme or confbridge, asterisk 1.2.x or Asterisk 10.x. Is it possible
to query a channel and get its conference number in return?
On Thu, Mar 22, 2012 at 11:09 AM, Satish Barot satish4aster...@gmail.comwrote:
Jayesh, Personally I haven't worked on Congbridge :).
Confbridge has evolved a
Hi there
How can I reset the value of asterisk' calls processed without restarting
asterisk? Where does it save/access the value of all processed calls since
last restart from?
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AFAIK:
Linux has a tendency to keep RAM filled up with any recently accessed
progarm. To keep programs access fast enough, it never removes something
from the memory, only replaces it, in case some program has more frequent
access than the one already present in ram.
If your server isn't
-users] Processed Call Counter
On Thu, 8 Mar 2012, [Digital^Dude] (r) wrote:
How can I reset the value of asterisk' calls processed without
restarting asterisk? Where does it save/access the value of all
processed calls since last restart from?
(I'm just a 1.2 Luddite, so my input may
by not stating
your Asterisk version or your technology (SIP/DAHDI/T1/etc). If you are
just using SIP trunks, SIP RELOAD might do it.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *[Digital^Dude] ®
*Sent:* Thursday
once made the
changes?
Leandro
2012/3/2 [Digital^Dude] ® millennium@gmail.com
I've tried with batch enabled as well as disabled, it seems irrespective
of the call burst I send to asterisk. CDR writes at a constant speed, not
changing with the call load!
On Fri, Mar 2, 2012 at 12:20 PM
Hello,
I'm using Asterisk 1.6.2.10. Whenever I dial Local channels via asterisk
manager, the calls never get a hangup signal even with timeout specified. I
find channels with ZOMBIE text appended.
It ends up occupying all the channels with the result that asterisk thinks
every channel is busy,
In almost all major releases of asterisk 1.6.x, SS7 Disposition never sets
to ANSWERED, even when someone answers the call, it logs NO ANSWER in
the cdrs.
Please help me resolve the issue.
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What versions on Asterisk and chan_ss7 are you using?
On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
I am experience the same issue.
Thanks
Vinod dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: [Digital^Dude
Are you using AMI originate for these SS7 outbound calls?
On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ®
millennium@gmail.comwrote:
What versions on Asterisk and chan_ss7 are you using?
On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
I am
--
*From: * [Digital^Dude] ® millennium@gmail.com
*Date: *Thu, 1 Mar 2012 18:23:47 +0500
*To: *vdharash...@gmail.com; Asterisk Users Mailing List -
Non-Commercial Discussionasterisk-users@lists.digium.com
*Subject: *Re: [asterisk-users] SS7 Disposition
Are you using AMI originate
Hi all,
It disturbs me to see asterisk (v 1.6.2.10) writing CDRs even when there
are 0 active channels and 0 active calls. Is there an upper limit in terms
of CDRs / second that asterisk can handle? Does it queue the unwritten CDRs
somewhere?
Please help me clarify this confusion.
Thanks
--
having to write continuosly in the
cdr backend. Writing in bunch instead one at once improves performance.
Check the cdr.conf file and disable the option batch if it hurts you.
Leandro
Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com
ha scritto
Hello,
I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source
file app_meetme.c is present in the apps dir. Also, I can find app_meetme
change-logs on the asterisk website. However, the dialplan doesn't have
this cmd. I have checked menuselect but it says it has been replaced
I did, and I mentioned it in my earlier email too.
Screenshot attached.
On Wed, Feb 22, 2012 at 6:03 PM, Doug Lytle supp...@drdos.info wrote:
[Digital^Dude] ® wrote:
I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its
source file app_meetme.c is present in the apps dir
Doug, I can find the following in asterisk 10 changelogs:
The following error will consistently
occur when trying to dial into a MeetMe conference when the
server does not have DAHDI hardware installed: app_meetme.c: No
DAHDI channel available for conference, user introduction
So you mean I can't use dahdi_dummy with meetme?
On Wed, Feb 22, 2012 at 9:28 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/22/2012 09:23 AM, [Digital^Dude] ® wrote:
Doug, I can find the following in asterisk 10 changelogs:
The following error will consistently
occur
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