Hi All,
Currently i'm facing with a cdr issue, When i originate a call (outbound
call) to uncorrect/unregistration user, asterisk inform me that call was
failed but in mysl-cdr (cdr-csv also) records.
Here are 2 samples
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Host
On Monday 08 April 2013, Thomas Perron wrote:
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Host
On Mon, Apr 8, 2013 at 4:26 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Monday 08 April 2013, Thomas Perron wrote:
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session
I don't think s extension will work on SIP channel. s extension is a
catch-all extension for Analog calls
Console output would be useful.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
On Mon, 8 Apr 2013, Thomas Perron wrote:
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
If you jack up logging, you may see a message on the console like:
looking for x in y
where x is the extension and y
Hello,
Many times, I need to test in a lab Asterisk servers before sending them to
customer locations.
I'm currently having trouble to test SIP trunks without touching SIP
configuration.
So, how should I change my testing lab so that I could now test SIP trunks
without modifying Asterisk server
2013-04-08 16:36, Olivier skrev:
Hello,
Many times, I need to test in a lab Asterisk servers before sending them
to customer locations.
I'm currently having trouble to test SIP trunks without touching SIP
configuration.
So, how should I change my testing lab so that I could now test SIP
trunks
2013/4/8 Johan Wilfer li...@jttech.se
2013-04-08 16:36, Olivier skrev:
Hello,
Many times, I need to test in a lab Asterisk servers before sending them
to customer locations.
I'm currently having trouble to test SIP trunks without touching SIP
configuration.
So, how should I change my
System details:
Digium Wildcard TDM410P with three extensions and one POTS line.
Dual core Pentium 4 (32-bit) processor
Fedora 18
Asterisk 11.2.1
DAHDI Version: 2.6.2 Echo Canceller: HWEC
I recently upgraded from Asterisk 1.4, and made as few changes to the
configuration files as possible. I
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