Did you check "ulimits" in Asterisk CLI?
Am 14.06.2012 16:02, schrieb [Digital^Dude] :
Hello,
Asterisk under
90% load of SS7 calls can only withstand the voice
broadcasting for 30 minutes. After around 30 minutes, it stops
Anybody, can you please share your thoughts to overcome this issue?
Hi,
I'm getting error: ' FAX session '9' is complete, result: 'FAILED'
(FAX_FAILURE_PARTIAL), error: '3RD_T2_TIMEOUT', pages: 1, resolution:
'204x196', transfer rate: '9600', remoteSID: '' ' when I tried send fax
more than
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
cat /etc/odbc.ini
--
[MySQL-asterisk]
Description = MySQL ODBC Driver
Driver = MySQL
Socket = /var/run/mysqld/mysqld.sock
Server =
- Original Message -
From: Thorsten Göllner t...@ovm-group.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 18, 2012 11:52:15 AM
Subject: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging
(mysql,
I think you need the DSN in car_odbr.ini to refer to the one in res_odbc.conf
as below
On 19/06/2012, at 3:52 AM, Thorsten Göllner wrote:
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql
database. But with no success. Do you have any hint for me?
cat
On 6/18/2012 11:52 AM, Thorsten Göllner wrote:
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
*SNIP*
But after a call hangup I get the following error:
cdr_odbc.c: Unable to retrieve database
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:
Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24
asterisk-sounds
1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1
(not trying to use the gui, want to do everything by hand) with a TDM410 with
2FXO and 2FXS. I have my POTS (PTNS) line plugged into
- 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 here
Hello,
a
On 06/17/2012 06:43 AM, Richard Kenner wrote:
Things work fine when he's talking to another Asterisk phone or to a SIP
trunk provider, but when connecting to a T1, there's clipping where about
1/3 of his voice (in intervals of maybe 200ms) are removed. This sounds
like an echo canceller
Matthew, thank you very much for the fast reply and very likely the
solution!
Using your hint I could locally reproduce the 488 Not Acceptable on my
Snom 300 with FW 8.7.3.7 by setting RTP encryption (SRTP) to on and
RTP/SAVP to off (RTP/SAVP mandatory or SRTP off - all fine). The person
calling
Hi,
Couple of moments ago my asteriskbox with a bri-card went down.
(burn-out)
I've heard that it seems to be possible to use an fritz!box as an
isdn-gateway (isdn -- sip)
Anyone around who has good/bad experiences with those AVM-boxes?
(yeah, i know it is tech overkill, but i'll get an
- 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 4:05:04 PM
Subject: Re: [asterisk-users] Error SIP/2.0 488 Not acceptable here
Matthew,
On Mon, Jun 4, 2012 at 4:11 PM, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
I have installed and configures this card in asterisk 1.6. When
trying to load the module codec_sangoma.so I see the following in
the asterisk log.
[2012-06-04 15:50:31] WARNING[18168]
On Wed, May 30, 2012 at 2:34 PM, Eric Wieling ewiel...@nyigc.com wrote:
Has anyone experienced an issue with Sangoma analog cards where the card
suddenly stops working? Trying to dial out shows the channel as busy, even
though there is no active call on that port?
I'd like to see the output
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