[asterisk-users] chan_ooh323 - cisco call manager express
Hello! I need to setup h323 trunk between cisco call manager express ( I have no access to it) and asterisk ( my side ). Calls from asterisk are OK, but there is no voice if calls are from cisco to asterisk. Looks like there is signalling problem. Could you , please look at https://issues.asterisk.org/jira/browse/ASTERISK-27138 and give me any suggestions? Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-Dial Handler return something like GOSUB_RESULT?
On Tue, Jul 18, 2017 at 6:49 PM, John Kiniston wrote: > I'm messing around with pre-dialer handlers today and running into a wall. > > Dial has the U option where I can execute a Gosub when the channels bridge > and there I can set the variable GOSUB_RESULT to BUSY to make Dial act like > the channel I called was Busy. > > I want to do something similar with a Pre-Dial handler but don't see a way > I can Set a variable or return a value that will cause Dial to act like the > channel I called was Busy? > > Use case: > Endpoint 100 calls Extension 101 > > Extension 101 has a Pre-Dial Handler that checks how many calls Endpoint > 101 has in progress and if it's greater than X returns a Busy. > > Dial acts like it got a Busy back from the Endpoint, Sets DIALSTATUS and > continues through it's dial-plan. > > I've tried using the BUSY() Application inside my Pre-Dial handler. > I've tried sending BUSY back as a Value with Return() to be picked up in > GOSUB_RETVAL > I've tried setting DIALSTATUS to BUSY. > > Am I trying to use the wrong tool for the Job here? > Why don't you do the how many calls the endpoint has check before Dial()? You can use the LOCK/UNLOCK functions as shown in [1] on the calling channel pre-dial routine to prevent reentrancy issues while doing the check. The called channel pre-dial routine is only to setup the channels you have decided to dial. > > Related, Why can we have multiple Hangup handlers but not Pre-Dial > handlers? > * There is only one dial to execute the called channel pre-dial handler while there are many opportunities to specify hangup handlers. * How do you think you could associate different pre-dial handlers to different called channels? Richard [1] http://blogs.asterisk.org/2017/03/29/dialplan-handler-routines-allow-customization/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pre-Dial Handler return something like GOSUB_RESULT?
I'm messing around with pre-dialer handlers today and running into a wall. Dial has the U option where I can execute a Gosub when the channels bridge and there I can set the variable GOSUB_RESULT to BUSY to make Dial act like the channel I called was Busy. I want to do something similar with a Pre-Dial handler but don't see a way I can Set a variable or return a value that will cause Dial to act like the channel I called was Busy? Use case: Endpoint 100 calls Extension 101 Extension 101 has a Pre-Dial Handler that checks how many calls Endpoint 101 has in progress and if it's greater than X returns a Busy. Dial acts like it got a Busy back from the Endpoint, Sets DIALSTATUS and continues through it's dial-plan. I've tried using the BUSY() Application inside my Pre-Dial handler. I've tried sending BUSY back as a Value with Return() to be picked up in GOSUB_RETVAL I've tried setting DIALSTATUS to BUSY. Am I trying to use the wrong tool for the Job here? Related, Why can we have multiple Hangup handlers but not Pre-Dial handlers? -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.16.0 segfault
I am getting frequent segfaults on a new Asterisk installation. So far the only message I see is: Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 7fb2d535723f sp 7fb25a11b5c0 error 4 in libasteriskpj.so.2[7fb2d52e5000+18] Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip 7f4afea0c23f sp 7f4a7f7e35c0 error 4 in libasteriskpj.so.2[7f4afe99a000+18] Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188 ip 7f2eb611923f sp 7f2e3aec25c0 error 4 in libasteriskpj.so.2[7f2eb60a7000+18] Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188 ip 7fc5758dd23f sp 7fc4fa6245c0 error 4 in libasteriskpj.so.2[7fc57586b000+18] Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188 ip 7f5f29abb23f sp 7f5eae8285c0 error 4 in libasteriskpj.so.2[7f5f29a49000+18] Since this is a Freepbx distro does could the problem be related to their flavor of Asterisk? I have several other plain Asterisk servers running on this version without any problems. Any recommendations on how to debug this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez dCAP #1349 +52 (55)8116-9161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users