I am sorry, all my fault ): I used click2dial and set a wrong outgoing line
- snom lines count there lines starting at 1 instead of 0. how could they
only dare to do so...
2012/10/3 Tim Nelson tnel...@rockbochs.com
- Original Message -
No idea? ):
How about showing your dialplan,
No idea? ):
2012/10/1 Stefan at WPF stefan.at@googlemail.com
Today I called some support hotline, for this support hotline no CDR was
created, also the call wasn't recorded, though there's a MixMonitor in my
dialplan, automatically recording every call.
Out of curiosity I set core set
Today I called some support hotline, for this support hotline no CDR was
created, also the call wasn't recorded, though there's a MixMonitor in my
dialplan, automatically recording every call.
Out of curiosity I set core set verbose 10 in the asterisk console. I
then dialed the support hotline
: Saturday, September 29, 2012 10:44:25 AM
Subject: Re: [asterisk-users] Reuse h extension?
On Sat, 29 Sep 2012, Stefan at WPF wrote:
I have 2 contexts, however both have the same h extension.
Currently I am doing copypaste for the h extension - is there a
better way?
Can I somehow
Actually, is there a difference between the callerid channel variable and
CDR(src)? Is CDR(src) actually set to the callerid channel variable? Thanks
:-)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
I have 2 contexts, however both have the same h extension.
Currently I am doing copypaste for the h extension - is there a better
way?
Can I somehow reference a h extension, so I have to create/modify it only
once?
Thanks for any hint!
--
How do you redirect all h calls to your status context? Thanks :-)
2012/9/29 Logan Bibby lo...@keobi.com
I have a status context with a hangup extension. All my h calls go
there.
- Logan
On Sep 29, 2012 4:32 AM, Stefan at WPF stefan.at@googlemail.com
wrote:
I have 2 contexts, however
, 2012 4:36 AM, Stefan at WPF stefan.at@googlemail.com
wrote:
How do you redirect all h calls to your status context? Thanks :-)
2012/9/29 Logan Bibby lo...@keobi.com
I have a status context with a hangup extension. All my h calls go
there.
- Logan
On Sep 29, 2012 4:32 AM, Stefan at WPF
In the cdr_custom.conf / cdr_sqlite3_custom.conf configuration files, is it
somehow possible to split the enumeration of CDR fields over multiple
lines? I get parsing errors when using more then a single line, but a
single line is very confusing if one has many CDR fields. So, is there
something
I am using Asterisk 1.8.10.1 on Ubuntu Server 12.04. I use MySQL to store
CDR records using cdr_adaptive_odbc.
The problem: When the MySQL Server fails for whatever reason, Asterisk
never reconnects automatically! So I loose all CDR informations even after
the MySQL server works again.
isql (used
I had problems on the Raspberry, like stuttering calls (just in between the
calls), maybe it was because of call recording but I would expect one call
recording to be not too much. However I used the packages from the repo,
maybe compiling it yourself and leaving out unnecessary stuff gives beter
?
Regards,
Qasim
On Tue, Sep 4, 2012 at 1:40 PM, Stefan at WPF
stefan.at@googlemail.com wrote:
I had problems on the Raspberry, like stuttering calls (just in between
the calls), maybe it was because of call recording but I would expect one
call recording to be not too much. However I
necessary stuff. You can exclude modules in runtime also and
it will serve you the same purpose.
Regards,
Qasim
On Tue, Sep 4, 2012 at 2:45 PM, Stefan at WPF
stefan.at@googlemail.com wrote:
Guess this is what most people are doing by compiling only necessary
stuff. Personally I
Anyone having any idea? Can post more details like my config etc. later,
but if someone has an idea now... let me know, thanks!
2012/8/31 Stefan at WPF stefan.at@googlemail.com
Thank you Doug, I tried that, but it doesn't have the expected effect. I
set it to 1 second and restarted
Doug, which kind of connection exactly are you using? I am using Adaptive
ODBC.
2012/9/3 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
Anyone having any idea? Can post more details like my config etc
With my Asterisk 10 setup, the initial connect may fail and then it does a
retry
Doug, if necessary I could change, just tell me, what you use, meaybe the
easiest solution ;-)
2012/9/3 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
Doug, which kind of connection exactly are you using? I am using Adaptive
ODBC.
Oh! I forgot you were using that. No, I'm not, sorry
Thanks, but which Asterisk module? (Iam also using UnixODBC by adaptive
odbc)
2012/9/3 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
just tell me, what you use, meaybe the easiest solution
UnixODBC
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty
Hello,
are there any deb packages for Ubuntu 12.04?
The repos at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packagesare
for older Ubuntu versions, also Asterisk 10 is only mentioned for YUM
/
CentOS?
Thanks :-)
--
_
--
Thank you Doug, I tried that, but it doesn't have the expected effect. I
set it to 1 second and restarted, no ODBC connection according to odbc
show. After a reload Asterisk is connected though.
2012/8/31 Doug Lytle supp...@drdos.info
So, how to tell Asterisk to automatically retry to connect
Hello all,
is it possible, to record calls directly as BLOB into a MySQL database?
Best regards
Stefan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
that into the blob.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stefan at WPF
*Sent:* Friday, August 31, 2012 10:22 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Record
Thank you very much Tim, this looks quite promising! Just sad, that once
again one has to compile it instead of provided packages ): But it's
probably worth the work :-)
2012/8/29 Tim Nelson tnel...@rockbochs.com
- Original Message -
Yeah, I noted that too, but besides that it seems
Thank you richard, now I understand and it works, except that all only gives
ssrc
themssrc
lp
rxjitter
rxcount
txjitter
txcount
rlp
rtt
and not all mentioned fields.
Do you know if there is a updated documentation anywhere?
I can guess what most of the fields are, but I am not
Looking at an older Channel documentation, it seems like the all
parameter wasn't adapted to the new, additional parameters. Will fill a bug
report during the day.
2012/8/25 Stefan at WPF stefan.at@googlemail.com
Thank you richard, now I understand and it works, except that all only
gives
Hello all,
I need some help understand the values of the CHANNEL function, e.g.
txploss // local packets loss
rxploss // remote packets loss
txjitter // local jitter
rxjitter // remote jitter
My main problem in understand is that a CHANNEL has two nodes (sender and
receiver), while a
. (I also would
find it more intuitive to have 2 legs and values for those instead of one
single value for 2 legs, but lets wait to find out, how to interpret those
things).
2012/8/25 Patrick Lists asterisk-l...@puzzled.xs4all.nl
On 25-08-12 14:31, Stefan at WPF wrote:
Hello all,
I need some
Using
exten = h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
gives me
[Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221
sip_acf_channel_read: Unrecognized argument
'rtpqos,audio,local_lostpackets' to CHANNEL
[Aug 24 12:08:10] WARNING[12087]: func_channel.c:393
A simply PHP based thing would be OK. Maybe I should look more specifically
for that or can anyone here recommend a PHP based CDR viewer?
Meanwhile I ended up building a mysql view, for private purposes it does
the job. A real solution would still be nice, though.
2012/8/23 Tim Nelson
If somebody is calling me using a wrong configured SIP phone, he gets back
an error message from my Asterisk server. That's ok, however I'd also like
to know that I missed a call. However there's no CDR entry created in that
case and checking the asterisk logs manually is not that great... Any way
Of *Stefan at WPF
*Sent:* Friday, August 24, 2012 7:43 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Log faulty calls?
** **
If somebody is calling me using a wrong configured SIP phone, he gets back
an error message from my Asterisk server. That's
/24 Eric Wieling ewiel...@nyigc.com
pbx*CLI core show function CHANNEL
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Friday, August 24, 2012 6:37 AM
To: Asterisk Users Mailing List
/display/AST/MSSQL+CDR+Backend
https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend
** **
Good luck!
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stefan at WPF
*Sent:* Monday, 20 August 2012 6:51
Hello,
just wondering if there is any easy to install CDR viewer? Easy meaning
install some package (debian system) and that's it. Had some problems
installing CDR-Stats, FreePBX also seems to be a longer task for setting
up. Isn't there a simple (productive :p) solution? Thanks :-)
Best regards
Hello all,
how can I verify if Asterisk is connected to a database using the ODBC
connection I configured?
According to
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html
(at
the end of the page) *odbc show on the console should output an
Doug, I have to admit I don't really understand your answer, except for the
maybe strange date ;-)
2012/8/19 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
how can I verify if Asterisk is connected to a database using the ODBC
connection I configured?
Mine shows:
odbc show
ODBC
, August 19, 2012 10:02 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] Verifying if Asterisk is connected using
ODBC?
Stefan at WPF wrote:
how can I verify if Asterisk is connected to a database using the ODBC
Hello,
I am currently trying to set up CDR logging. I got all the ODBC stuff for
my mysql server set up, also checked there's a connection using odbc show
in the asterisk console:
Name: asterisk
DSN:asterisk-connector
Last connection attempt: 1970-01-01 01:00:00
Pooled: No
:
[general]
enable = yes
unanswered = yes
2012/8/19 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
Thanks for any hint :-)
/etc/odbc.ini
[MySQL-cdr]
Description = CDR Database
Driver = MySQL
Socket = /var/lib/mysql/mysql.sock
Server = 172.19.8.9
User
AM, Stefan at WPF wrote:
Hmm is it possible, that the monitor command changes the quality? If not
I guess I also once have to try compiling it from source, though I
wanted to avoid that.
It certainly can, since recording the call causes disk I/O as the audio is
written out. In addition
Hmm is it possible, that the monitor command changes the quality? If not I
guess I also once have to try compiling it from source, though I wanted to
avoid that.
2012/7/23 Bakko asannu...@gmail.com
Hello,
I tried Asterisk Confbridge with raspberry pi without audio issue.
Asterisk was
:13 PM, Doug Lytle wrote:
Stefan at WPF wrote:
Snom 300 - Asterisk 1.8.13.0 running on Raspberry Pi
Not that I can help, but I'm sorta shocked that you have Asterisk running
on a Raspberry Pi!
I've had it running on a Guruplug without any problems - http://www
Hello,
I am currently using the following setup:
Snom 300 - Asterisk 1.8.13.0 running on Raspberry Pi - Sipgate SIP Provider
When I am using this setup, the call quality isn't as good as when using a
direct connection like
Snom 300 - Sipgate SIP Provider
to my SIP Provider (Sipgate).
Sipgate
Hello Bruce,
2012/6/22 Bruce B bruceb...@gmail.com
Thanks. Want to secure everything and anything possible.
1- Can both SIP over TLS and SRTP work in conjunction to each other?
yes, you even need them both. SRTP encrypts just the media (audio), but the
encryption key for it has to be sent
Hello,
1) I am wondering what is the best practice to monitor if there are or were
problems with SIP calls on my Asterisk box. E.g. how about a software that
extracts all calls from the /var/log/asterisk/full (I have permanently
enabled verbose 10 and sip debug) log and tells me on which of them
Hello,
is there anywhere an overview of SIP error codes and under which condition
they are reported by Asterisk?
There are general definitions for SIP error codes, but they are quite
general and it's Asterisk that actually checks what's wrong and then
reports an error. Now, currently I could
Yeah, I noted that too, but besides that it seems like it is exactly what I
am looking for. I am especially confused that there's no hint like hey,
buy our new product, just EOL. So let's say I am looking for an
alternative to this. And unfortunately I have to add it's for private use
and I
. Describing these concrete occurences of errors is what I am
currently missing.
2012/6/20 Jonathan Rose jr...@digium.com
Stefan at WPF wrote:
is there anywhere an overview of SIP error codes and under which
condition they are reported by Asterisk?
There are general definitions for SIP error
Hello,
a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
mjor...@digium.com
- Original Message -
From: Stefan at WPF stefan.at@googlemail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 18, 2012 3:04:32 PM
Subject: [asterisk-users] Error SIP/2.0 488 Not acceptable
Ah, so, really easy ;-) Thank you very much for this :-)
2012/6/17 Warren Selby wcse...@selbytech.com
On Sat, Jun 16, 2012 at 4:23 PM, Stefan at WPF
stefan.at@googlemail.com wrote:
Hello,
I have an internal extension, e.g. 1005 which is being called from an
external/public number
Hello,
I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like number 12345 not available I was only
hearing 345 not available. Verbose level 5 on the asterisk console didn't
to a pause of
some seconds.
2) Your solutions handles the symptoms of the problem, I'd like to fix the
root cause of this problem.
Any ideas on number 2, fixing / finding the root cause of this problem?
Thanks :-)
2012/6/17 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
beginning
2012/6/17 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
2) Your solutions handles the symptoms of the problem, I'd like to fix
the root cause of this problem.
The root cause of the problem (Most likely) is that the channel hadn't be
answered. A wait, allows the channel
that the
problem is on the mobile phone side?
2012/6/17 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
Which end do you mean with channel not answered? The asterisk
The Asterisk side. If the answer didn't fix the issue, then my guess is
that it's on the cellular provider's side (Which isn't unheard
prompt and where you only hear part of the
prompt.
Also, if you can clarify the infrastructure setup as well, that would be
helpful.
Thanks,
--Warren Selby, dCAP
On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com
wrote:
Hmm, I tried calling myself (the asterisk
Debian does not) running on a Raspberry Pi http://www.raspberrypi.org/:-)
2012/6/17 Stefan at WPF stefan.at@googlemail.com
Thank you Warren,
I will temporarily skip this step, as I don't have the problem anymore,
though I don't know why (for that and learning purposes the logs maybe
would
Hello,
I have an internal extension, e.g. 1005 which is being called from an
external/public number like 123456789. Now when it comes to the spoken
voicemail information it says something like number 1000 not available,
however it should say number 123456789 not available. How can I configure
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