We have just implemented cdr-custom. Works fine minus the timestamps that
appear in the CDR.
The system's timezone is GMT. In the configuration usegmtime=yes is set.
However, all of the CDRs in the Custom CDR comes as GMT-5.
Another oddity is that the standard cdr/Master.csv is using
Be sure your OS79XX.TXT and SIPDefault.cnf file and SIP[MACADDRESS].cnf file
all agree on the version of software the phones are to be running.
For example OS79XX.TXT should read: P0S3-08-2-00, and in SIPDefault.cnf a
line should read: image_version:P0S3-08-2-00. If you were trying to run
I have been forced to introduce a Cisco 7971G-GE into my
network, because it has a pretty screen. I have wasted
nearly three days fighting with the thing based upon the information on
voip-info.org and a few other forums.
Asterisk is reporting a 401 Unauthorized. Which
typically means
If Asterisk was used to set up and tear down calls, and
using canreinvite allowing the RTP to pass from end-point to end-point, how
many calls could Asterisk handle at once?
I ask because I have been utilizing OpenSER but find myself constantly
needing Asterisk to do this or that, and
Has anyone ever published a concise howto or good documentation on how the two interrelate? and Configurations.. On 6/13/06, BILL GITONGA
[EMAIL PROTECTED] wrote:Asterisk does to scale well. Use OpenSER or SER as a
front end to asterisk. Make all the sip traffic gothrough ser and only go to
We are using Asterisk in a purely VOIP environment, on leased dedicated server at a dedicated server provider. It is becoming more and more apparent that this dedicated server is actually a vritualized server.
We have now found a need to utilize the MeetMe application for conferencing. However we
I am monitoring via my queues.conf.
[310]
wrapuptime=30
timeout=15
strategy=ringall
retry=5
queue-youarenext=
queue-thereare=
queue-thankyou=custom/aa_6
queue-callswaiting=
music=Support
monitor-join=yes
monitor-format=gsm
maxlen=0
leavewhenempty=no
joinempty=no
context=aa_6
announce-holdtime=no
I have an ongoing problem and do not know where to begin
troubleshooting it. We run a helpdesk, and call recording is
extremely important. But we have found that calls are recorded at
random. We receive the call via our toll-free number over an IAX
connection. The call is then either handled by a
We have setup an Asterisk server and everything works great
with the exception of Music on Hold.
If you dial into our system and are placed in a Queue, you
get music. If you are placed on Hold no music (which I believe may be
caused by the XPro), or if you are parked you get no music.