Re: [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device
On Aug 16, 2012, at 6:25 PM, Tiago Geada wrote: forward to a Local extension that has dialplan requiring the acknowledgement? On 16 August 2012 21:12, Phil Frost p...@macprofessionals.com wrote: I'd like to allow my users to forward their calls using the forwarding feature on their SIP handsets and continue to receive Queue() calls. Currently I set the 'i' option in Queue() so that if a user forwards to their cell phone, or any other extension that has voicemail, the voicemail doesn't eat all the calls to the queue. I'd think that would require teaching all the users to forward to a different extension if they thought they could be receiving queue calls. My users probably aren't that good at following directions ;) Ultimately, I'm sure I could solve this problem by taking management of forwarding off the phone and into Asterisk, since then I'd absolutely have some flag indicating if forwarding is active or not. However, I was just hoping there was an easier way. I'm really happy with the forwarding interface on our current handsets, and I'd rather not go through the effort of changing their configuration, or changing the user experience if I can avoid it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available - why not?
On Mar 28, 2012, at 4:31 PM, Phil Frost wrote: I'm attempting to direct my queue logs at a PostgreSQL table, and seeing this error in the asterisk console: config.c:2256 find_engine: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available However, everything I know how to check indicates the odbc engine is available. I can't find any more verbose description of the error from Asterisk, so I'm unsure how to proceed. Well, after making a debug build and stepping through the source I solved it. It wasn't apparent to this neophyte that there's res_odbc.so, and then there's res_*config*_odbc.so, which is set to noload in the default modules.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Mac Address on connected IP phones
On Mar 13, 2012, at 5:51 PM, resea...@businesstz.com wrote: I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Your router knows the MAC addresses of the phones. So does your DHCP server, if they are using DHCP. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NOT in the media path
On Feb 29, 2012, at 7:25 AM, Jonas Kellens wrote: The Asterisk server still stays in the SIP Signaling path of the call, just media does not flow through the server. You can verify this by running a SIP debug and looking at the media endpoints. What is it that I should be looking for in the SIP debug information ? Is it in the SDP-body ? To set up direct media, Asterisk will send a re-invite with an SDP body containing the address of the other endpoint. RTP then flows directly between endpoints. If you just want to know if Asterisk is in the media path or not, you can also use rtp set debug ... and Asterisk will log a line for each RTP packet it handles. -- v: 248.893.0738 | f: 248.893.0747 http://macprofessionals.com/ find us on facebookall -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
