Re: [asterisk-users] sip show channels - gives a growing list of dead channels
Asterisk SIP channels can hang for a variety of reasons such as network errors, signaling malfunction and software bugs. These are difficult to track down and sometimes the root cause is not even in your control. In order to provide a sort of garbage collection mechanism for such hung SIP channels, Asterisk 1.6 supports a mechanism called as SIP Session Timers. You may want to give this feature a shot. The instructions for configuring it are in sip.conf. -- Raj On Mon, Mar 10, 2008 at 5:13 PM, Keith Hardee [EMAIL PROTECTED] wrote: I feel like I've seen that error before, but I did some quick testing and was not able to produce the error. CLI level was greater than 206 (many v's) callfromto hangup Test 1polycom spectralink polycom Test 2polycom spectralink spectralink Test 3spectralink polycom polycom Test 4spectralink polycom spectralink Test 5 spectralink spectralink spectralink I only did one test of each above because I am not in office (had someone doing tests while I watched CLI). I can test more when I get back Thursday. Thanks for input. On Sat, Mar 8, 2008 at 2:59 AM, Fons van der Beek [EMAIL PROTECTED] wrote: Same problem over here I use KIRK-Telecom ip600v3 This only happens on calls between SIP en MiSDN, anyone any clue? As far as i can see these dead calls once in while occur when the remote party first hangs up (remote=MiSDN channel) Keith do you also have error messages in the CLI when you open asterisk by using asterisk -rvv ? (a lot of v) -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 10.0.0.71 represents the IP number of internal phone Keith Hardee schreef: I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 7fa08c964a1 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 542 7088c6a7-db 00102/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 424cc109052 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 225fe5130e5 00104/00102 0x0 (nothing) No Rx: BYE Thanks, Keith ___ -- 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 ___ -- 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 ___ -- 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 -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org ___ -- 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] sip show channels - gives a growing list of dead channels
I feel like I've seen that error before, but I did some quick testing and was not able to produce the error. CLI level was greater than 206 (many v's) callfromto hangup Test 1polycom spectralink polycom Test 2polycom spectralink spectralink Test 3spectralink polycom polycom Test 4spectralink polycom spectralink Test 5 spectralink spectralink spectralink I only did one test of each above because I am not in office (had someone doing tests while I watched CLI). I can test more when I get back Thursday. Thanks for input. On Sat, Mar 8, 2008 at 2:59 AM, Fons van der Beek [EMAIL PROTECTED] wrote: Same problem over here I use KIRK-Telecom ip600v3 This only happens on calls between SIP en MiSDN, anyone any clue? As far as i can see these dead calls once in while occur when the remote party first hangs up (remote=MiSDN channel) Keith do you also have error messages in the CLI when you open asterisk by using asterisk -rvv ? (a lot of v) -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 10.0.0.71 represents the IP number of internal phone Keith Hardee schreef: I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 7fa08c964a1 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 542 7088c6a7-db 00102/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 424cc109052 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 225fe5130e5 00104/00102 0x0 (nothing) No Rx: BYE Thanks, Keith ___ -- 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 ___ -- 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 ___ -- 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
[asterisk-users] sip show channels - gives a growing list of dead channels
I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 7fa08c964a1 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 542 7088c6a7-db 00102/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 424cc109052 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 225fe5130e5 00104/00102 0x0 (nothing) No Rx: BYE Thanks, Keith ___ -- 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] sip show channels - gives a growing list of dead channels
Same problem over here I use KIRK-Telecom ip600v3 This only happens on calls between SIP en MiSDN, anyone any clue? As far as i can see these dead calls once in while occur when the remote party first hangs up (remote=MiSDN channel) Keith do you also have error messages in the CLI when you open asterisk by using asterisk -rvv ? (a lot of v) -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 10.0.0.71 represents the IP number of internal phone Keith Hardee schreef: I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 7fa08c964a1 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 542 7088c6a7-db 00102/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 424cc109052 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 225fe5130e5 00104/00102 0x0 (nothing) No Rx: BYE Thanks, Keith ___ -- 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 ___ -- 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
