[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 GMT. Please advise. Thanks, kw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 P003-08-2-00 The image_version line is not required in SIP[MACADDRESS].cnf file so I wouldn't put it there otherwise if you have many phones you'll have to edit each phone each time you change the software. kw _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh Sent: Sunday, June 17, 2007 8:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Upgrade cisco SIP phone 7940 Hi All, My current 7940 phones use P0S3-06-3-00. I'd like to upgrade them so they're not massively out of date. I found a page at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx that gives some info, and using the cisco links there have tried to upgrade. According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades again and reboots, then the cycle starts again. Anyone had any success in doing this? Adrian Marsh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 bad username/password combination. Unfortunately, all of the usernames and passwords I see in the config file, are as they appear in the sip.conf. Is it possible to get further debug information from Asterisk regarding the 401ie. What Asterisk doesnt like about it? The chances that the information I have obtained from voip-info.org or the other forums that discuss the 7971G-GE and Asterisk could be incomplete. Would anyone happen to have a working configuration for the 7971G-GE (running SIP70.8-0-4SR1S) they would care to share, or allow me to purchase. I really need to get the child who signs my paycheck his pretty screen. Thanks in advance, kw ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 would like to move OpenSER into more of a Registration server, and letting Asterisk handle all of my calls I understand that the set up and tear down may be a tad slower, but programming (using AGI, etc.) would definitely outweigh the timing IMO. Thanks in advance. Kw ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 for voicemail, IVRi.e media stuff. If you connect to the PSTN using sip,then SER would be used for routing all PSTN calls. --- Erick Perez [EMAIL PROTECTED] wrote: While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same box as asterisk or separated? does it add more functions to asterisk? or is the main function to better handle SIP over firewalls (due to SIP over TCP support)? Thanks for the explanation. -- Erick Perez ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 have no Zaptel hardware. We have attempted to build the ztdummy kernel module for the server but are finding ourselves unable to do so because we do not have the kernel source on the box (as the dedicated server provider does not make it available, and typical resources for kernel sources causes the dedicated server to crash). In short, does anyone have any other advice to get the MeetMe application working on a potentially virtualized server (although the box said dedicated), without kernel sources, and a box that has no apparent USB? Thank you so much in advance for your advice.kw ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 announce-frequency=60 On 11/16/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Are you doing agent recording, or recording through the dialplan ?JulianKelvin Williams wrote: 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 local extension or sent outside through another phone line. As I said, it records at random--whether the call is answered both internally or connected to another PSTN.The following two calls (taken from the queue log) were several minutes in length, however, it shows it being completed immediately. 1132104643|1132104620.0|310|NONE|ENTERQUEUE||Roselle IL REMOVED 1132104667|1132104620.0|310|Local/[EMAIL PROTECTED],1|CONNECT|24 1132104667|1132104620.0|310|Local/[EMAIL PROTECTED] ,1|COMPLETECALLER|24|0 1132105549|1132105526.6|310|NONE|ENTERQUEUE||Haufman L. REMOVED 1132105581|1132105526.6|310|Local/[EMAIL PROTECTED],1|CONNECT|32 1132105581|1132105526.6|310|Local/[EMAIL PROTECTED] ,1|COMPLETECALLER|32|0 Where can I fix this? Any help is greatly appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 local extension or sent outside through another phone line. As I said, it records at random--whether the call is answered both internally or connected to another PSTN.The following two calls (taken from the queue log) were several minutes in length, however, it shows it being completed immediately. 1132104643|1132104620.0|310|NONE|ENTERQUEUE||Roselle IL REMOVED 1132104667|1132104620.0|310|Local/[EMAIL PROTECTED],1|CONNECT|24 1132104667|1132104620.0|310|Local/[EMAIL PROTECTED],1|COMPLETECALLER|24|0 1132105549|1132105526.6|310|NONE|ENTERQUEUE||Haufman L. REMOVED 1132105581|1132105526.6|310|Local/[EMAIL PROTECTED],1|CONNECT|32 1132105581|1132105526.6|310|Local/[EMAIL PROTECTED],1|COMPLETECALLER|32|0 Where can I fix this? Any help is greatly appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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. I can understand the hold (if it is truly held by the client only) but the Park should still get music. Im attaching the relevant confs if someone sees a problem please advise. [EMAIL PROTECTED] asterisk]# cat musiconhold.conf ; ; Music on hold class definitions ; #include musiconhold_additional.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z [EMAIL PROTECTED] asterisk]# cat musiconhold_additional.conf [classes] NOC = quietmp3:/var/lib/asterisk/mohmp3/NOC Support = quietmp3:/var/lib/asterisk/mohmp3/Support [EMAIL PROTECTED] asterisk]# cat queues.conf [general] ; ; Global settings for call queues ; (none exist currently) ; ; Note that a timeout to fail out of a queue may be passed as part of application call ; from extensions.conf: ; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) ; example: Queue(dave|t|||45) [default] ; ; Default settings for queues (currently unused) ; #include queues_custom.conf #include queues_additional.conf [EMAIL PROTECTED] asterisk]# cat queues_additional.conf [301] wrapuptime=0 timeout=15 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=custom/aa_3 queue-callswaiting=queue-callswaiting music=Support monitor-join=yes monitor-format=gsm maxlen=0 leavewhenempty=yes joinempty=no context=aa_3 announce-holdtime=yes announce-frequency=90 [302] wrapuptime=0 timeout=15 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=custom/aa_3 queue-callswaiting=queue-callswaiting music=Support monitor-join=yes monitor-format=gsm maxlen=0 leavewhenempty=no joinempty=yes context=aa_3 announce-holdtime=yes announce-frequency=90 [303] wrapuptime=0 timeout=15 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=NOC monitor-join=yes monitor-format= maxlen=0 leavewhenempty=no joinempty=yes context= announce-holdtime=yes announce-frequency=60 [304] wrapuptime=0 timeout=15 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=NOC monitor-join=yes monitor-format=gsm maxlen=0 leavewhenempty=no joinempty=yes context= announce-holdtime=yes announce-frequency=60 [EMAIL PROTECTED] asterisk]# ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
