[asterisk-users] Asterisk and QSIG
Hello, I'm looking for info about interconnecting asterisk to QSIG GF enabled PABX over PRI . Any information and pointers will be helpful. The very first first question: does asterisk support QSIG BC and GF natively i see that it is supported through CAPI enabled cards but what about support through librip/dahdi? Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Invite somebody to a conf call
Hello, I wonder if somebody can help me with following: I need to acheive something similar to this: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO but with a twist. Suppose i'm already in a meetme conf call i want to dial a * for example, hear the dial tone dial the destination number which will be bridged to the original conf call and return back myself to the original conf call Secondary issue: How to do the same but from the dynamic conf call created by Page application Thanks Vadim ___ -- 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] Timeout question
I wonder which timeout will apply here: the one in master context or one from the slave context? [master] exten=>100,1,Dial(Local/[EMAIL PROTECTED], 20) [slave] exten=>100,1,Dial(SIP/100, 30) Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange putconfig bahviour
Vadim Lebedev mbdsys.com> writes: I've found the root of the problem and fixed it: http://bugs.digium.com/view.php?id=13341 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange putconfig bahviour
> Why is it bad? In all Asterisk config files, the '>' after the '=' is > superfluous for defining extensions, variables, etc. Try it. Having > exten=123,1,... is perfectly valid and does not affect how Asterisk works > in any appreciable way. > Ok Tighlman, Thank you for the information, i didn't know about is as ALL asterisk docs says that you're supposed to use exten=> format. So i did try putconfig without '>' chars and have the same problem: = Action: updateconfig reload: no SrcFilename: extensions.conf DstFilename: extensions.conf.new Action-00: newcat Cat-00: ami-test-01 Action-01: append Var-01: exten Value-01: 888,1,Noop(888) Cat-01: ami-test-01 Action-02: append Var-02: exten Value-02: 999,1,Noop(999) Cat-02: ami-test-0 Response: Error Message: Save of config failed Any ideas on wher to look for the reasons or how to debug Tanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange putconfig bahviour
> > Value-01: >888,1,Noop(888) > > It may or may not be related, but those greater-than signs (>) in there > look wrong, and may be causing some problems. > Without them you can't get "exten=>888,1,Dial(888)" if your remove '>' you get "exten=888,1,Dial(888)" in extensions.conf which is BAD. And anyway as i've said it wokr ok on one of the atsreisk and fails on second one. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange putconfig bahviour
Hello, I'm running identical putconfig AMI action against 2 asterisks == Action: updateconfig reload: no SrcFilename: extensions.conf DstFilename: extensions.conf.new Action-00: newcat Cat-00: ami-test-01 Action-01: append Var-01: exten Value-01: >888,1,Noop(888) Cat-01: ami-test-01 Action-02: append Var-02: exten Value-02: >999,1,Noop(999) Cat-02: ami-test-01 == It works ok on one server and returns error on the second: Response: Error Message: Save of config failed I did check for obvious things like permissions in manager.conf Where i should be looking to resolve this issue? Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI and extensions.conf
> > > > Ok i've fixed the problem (actually there was two of them > > 1) space in after colon in "dstfilename: extensions.conf" and > > Does this really matter? > I know it is strange, but yes, Once i've fixed it in the first request it start to add a new category to the file > > 2) numiric id have to be in XX (6 digit) format > > Is there a simple way to find "incorrect headers" in manager messages? > e.g.: a debug message about "header Action- does nothing"? I saw no debug messages Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI and extensions.conf
Vadim Lebedev mbdsys.com> writes: > > Tzafrir Cohen xorcom.com> writes: > > > > > On Thu, Aug 14, 2008 at 04:06:27PM +, Vadim Lebedev wrote: > > > Hello > > > > > > I'm looking for a wayy to modify extensions.conf > > > It seems that PutConfig AMI command is not > > > supposed to work on extensionsq.conf > > > > It should. Do you have a test case where it doesn't? > > > Action: updateconfig > reload: no > srcfilename: extensions.conf > dstfilename: extensions.conf > Action-: append > Var-: exten > Value-: 999,1,Noop(999) > Cat-: ami-test > > Response: Success > > But the file itself is not modified > > ___ Ok i've fixed the problem (actually there was two of them 1) space in after colon in "dstfilename: extensions.conf" and 2) numiric id have to be in XX (6 digit) format I've also fixed text here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+UpdateConfig which erronously stated that thi command is unable to handle extensions.conf agents.conf , etc (where you have var => value form) Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI and extensions.conf
Tzafrir Cohen xorcom.com> writes: > > On Thu, Aug 14, 2008 at 04:06:27PM +0000, Vadim Lebedev wrote: > > Hello > > > > I'm looking for a wayy to modify extensions.conf > > It seems that PutConfig AMI command is not > > supposed to work on extensionsq.conf > > It should. Do you have a test case where it doesn't? > Look at this trace: Action: login Username: castel Secret: castel Asterisk Call Manager/1.0 Response: Success Message: Authentication accepted Action: updateconfig reload: no srcfilename: extensions.conf dstfilename: extensions.conf Action-: newcat Cat-: ami-test Response: Success Action: updateconfig reload: no srcfilename: extensions.conf dstfilename: extensions.conf Action-: append Var-: exten Value-: 999,1,Noop(999) Cat-: ami-test Response: Success But the file itself is not modified ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI and extensions.conf
Hello I'm looking for a wayy to modify extensions.conf It seems that PutConfig AMI command is not supposed to work on extensionsq.conf Any ideas? Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Browser based VoIP client?
Hilary Miller gmail.com> writes: > > Something that I can put on our internal company website to replace > our hardware IP phones. > > I see many web 2.0 startups offering browser based clients for their > own service, but I can't seem to find anything that I can use with my > own PBX. Do I suck at searching google or has the future not arrived > yet? > > Thanks for reading! Take a look at http://www.mbdsys.com/opensource/veronix Vadim ___ -- 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] DUNDi question
Hello, I'm wondering about following DUNDI setup Suppose we have 2 Asterisks: astA and astB with DUNDI peering active between them and 2 SIP endpoints: sipA registered with astA and sipB regsitered on astB All this is on the same LAN now sipA call an number which corresponds to [EMAIL PROTECTED] , so astA lookups thru DUNDI at astB and forwards the call there. My question is how this fowarding is done ? Using SIP RE-INVITE, or REFER, or using SIP 301 responce with Contact pointing at [EMAIL PROTECTED] And does the final RTP stream traverse both Asterisks or only one of them or None of them? Thanks Vadim ___ -- 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
