Hi Team,

I would like capture SS7 Error Code in CDRs, Specifically of outbound call
from the asterisk. calls generated using .call file.

In extension.conf extens gets excuted on successful call only , So that on
h extension reason of hangup is captured. But i am not aware of any
provision that capture on Unsuccessful call.

please guide on this or suggest any patch.

Thanks
Vinod d

Hi

I was asked by our development departement to setup asterisk in a
manner that if someone calls an extension in the department that was
was only configured, but a handset was never attached to it to fall
back to a default extension. For example: Someone calls extension
2408, but there’s no phone attached to 2408 it should fall back and
ring at 2400..

How do I setup asterisk to find out if there’s a phone attached to an
internal number if not ring another extension?

TIA
Paolo

I’ve got an Asterisk 10.1.2 server using res_odbc to make a connection to a MSSQL server on a different machine for a timeclock extension we have running. The connection works just fine until Asterisk’s network path to the MSSQL server gets disrupted (for whatever reason), and then all future attempts to contact the MSSQL server cause the timeclock call to hang, until Asterisk is completely restarted. Trying to module unload/load res_odbc.so crashes Asterisk. I’ve been using the following config, and tried with preconnect both on and off:

[timeclock]
enabled => yes
dsn => timeclock
username => notauser
password => stealme
pooling => no
limit => 1
pre-connect => no

Any suggestions on where to try digging in to this?

Thank you,

Noah Engelberth
MetaLINK Technologies

The message does not contain any threats
AVG for MS Exchange Server (2012.0.1913 – 2114/4882)

Hi,

I would like to test the following COLP use case :

Alice and Bob are both using a SIP phone registered on a Asterisk 10 server.
Alice dials Bob’s extension.
While Bob’s phone is ringing, Asterisk updates Alice phone screen with
Bob’s name, so that at a glance, Alice can check she dialed the
correct number.

Before diving into Asterisk documentation, I would be happy to be
confirmed if one of the following SIP phone support this feature :

Aastra 57i
Yealink T26
Cisco 525G
Thomson ST2030S

Regards

Hi All;

Is there an admin tool (web based) to check the voicemail and manipulate it (delete all the voicemail under extension, showing how many voicemails for the extension, … etc)?

Can AsteriskNow do this? Or any other recommended tool that it is very good in this?

Regards
Bilal