Re: [asterisk-users] 40sec between dial execution and sending SIP request
thanks, this delay is occurred on asterisk server, between dial execution and CALLED . On Mon, May 9, 2011 at 7:12 PM, Warren Selby wcse...@selbytech.com wrote: On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net wrote: Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Check the dial timeout on your phone itself. What model phone do you have? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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 -- Pezhman Lali -- _ -- 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] 40sec between dial execution and sending SIP request
Show us the cli trace of the delay. Thanks, --Warren Selby, dCAP On May 10, 2011, at 2:18 AM, Pezhman Lali l...@lopl.net wrote: thanks, this delay is occurred on asterisk server, between dial execution and CALLED . On Mon, May 9, 2011 at 7:12 PM, Warren Selby wcse...@selbytech.com wrote: On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net wrote: Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Check the dial timeout on your phone itself. What model phone do you have? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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 -- Pezhman Lali -- _ -- 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 -- _ -- 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] 40sec between dial execution and sending SIP request
Good call Warren, might I add that a great idea would be to set debug and verbose to 5, change the timestamp format on your logs temporarily to show HH:mm:ss:ms (don't necessarily need milliseconds, but I'm an accuracy geek), make sure you have a log that is writing ALL output (except maybe DTMF, but error, warning, info, debug, verbose are all necessary) then do a logger reload and a logger rotate, dial your test call, and then attach the resulting logfile. On Tue, May 10, 2011 at 2:28 AM, Warren Selby wcse...@selbytech.com wrote: Show us the cli trace of the delay. Thanks, --Warren Selby, dCAP On May 10, 2011, at 2:18 AM, Pezhman Lali l...@lopl.net wrote: thanks, this delay is occurred on asterisk server, between dial execution and CALLED . On Mon, May 9, 2011 at 7:12 PM, Warren Selby wcse...@selbytech.com wcse...@selbytech.com wrote: On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net l...@lopl.net wrote: Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Check the dial timeout on your phone itself. What model phone do you have? -- Thanks, --Warren Selby, dCAP http://www.selbytech.comhttp://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Pezhman Lali -- _ -- 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- 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] 40sec between dial execution and sending SIP request
Dears thanks for your all helps. with assigning ip to the sip.conf and disabling srv_lookup the delay was removed. thanks again best On Tue, May 10, 2011 at 12:31 PM, mahesh katta maheshka...@flexydial.comwrote: Dear Pezhman Lali, Just below lines add in you sip.conf, after this if you get same problem do 2nd step, that is run on your server update_server_ip command this for your database is not matching the your current IP.. externip=abc.net.org(if your server access remotly that URL add here if not leave that line ) localnet=10.10.10.0/255.255.255.0(your local ip address series) On Mon, May 9, 2011 at 5:56 PM, Pezhman Lali l...@lopl.net wrote: Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Best regards -- Pezhman Lali -- _ -- 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 -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- 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 -- Pezhman Lali -- _ -- 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
[asterisk-users] 40sec between dial execution and sending SIP request
Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Best regards -- Pezhman Lali -- _ -- 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] 40sec between dial execution and sending SIP request
On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net wrote: Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Check the dial timeout on your phone itself. What model phone do you have? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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
