[asterisk-users] CDR Timestamps (cdr-custom)

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

Re: [asterisk-users] Upgrade cisco SIP phone 7940

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

[asterisk-users] Asterisk Performance without RTP?

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

[Asterisk-Users] Music On Hold Problem

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.

[Asterisk-Users] MeetMe/Asterisk Timer

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

Re: [Asterisk-Users] OPENSER / SER and Asterisk

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

[Asterisk-Users] Queue Monitoring..

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

Re: [Asterisk-Users] Queue Monitoring..

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

[asterisk-users] Cisco 7971G-GE SEP{MAC}.cnf.xml

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