[asterisk-users] Permanent sip and agi debug on?
Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Licensing question.
On Tuesday 08 November 2011, Yaroslav Panych wrote: Greetings I have found next paragraph in Licence file(source root) Digium, Inc. (formerly Linux Support Services) holds copyright and/or sufficient licenses to all components of the Asterisk package, and therefore can grant, at its sole discretion, the ability for companies, individuals, or organizations to create proprietary or Open Source (even if not GPL) modules which may be dynamically linked at runtime with the portions of Asterisk which fall under our copyright/license umbrella, or are distributed under more flexible licenses than GPL. What does it mean? Does it mean I can write non-GPL modules(BSD, MIT, etc)? Can I build my modules in common asterisk source tree(i.e. using LOCAL_MOD_SUBDIRS=my_mod_subdirs_list make ) or must use separate tree? If so, then since Asterisk core does not accepts anything except AST_MODULE_INFO(ASTERISK_GPL_KEY, ) what I should do here? If you write modules that need to be compiled against the Asterisk Source Code, then the resulting compiled binaries are by definition derivative works of Asterisk. The GPL already gives you permission to release those modules under the GPL. And Fair Dealing / Fair Use provisions of copyright law mean you need no explicit permission to make use of those modules yourself for their rightful purpose. You require a special, separate licence from Asterisk to distribute compiled binaries which are derived works of Asterisk under anything but the GPL. Other people can, also under Fair Dealing provisions, compile their own legitimately-acquired copy of your module Source Code against their own legitimately-acquired copy of the Asterisk Source Code; but what they end up with may well be unredistributable. What you *can't* do is distribute your modules *as pre-compiled binaries* under any licence beside the GPL -- if they are distributed under any other licence, they *must* be compiled on-site by the end user. You've been given the Asterisk Source Code freely, in the hope that it will be useful to you. The *least* you can do is share any improvements you may make with the rest of the world, on the same terms as the rest of the world shared Asterisk with you in the first place so you could make those improvements. Shorter version: Leeches not welcome. -- AJS Answers come *after* questions. -- _ -- 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] No call progress sounds
On Nov 8, 2011, at 9:55 AM, isr...@gmail.com wrote: There is a bug which blocks call progress message 8 which was fixed but I don't remember in which version Try upgrading to latest 1.6 version Before we opened for the day today I updated to 1.6.2.20 and that seems to have solved the call progress problem. Thanks -chris www.mythtech.net -- _ -- 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] State of Asterisk+Virtualization+Timing
Is anybody using pci-passthrough? 2011/11/9 Nick Khamis sym...@gmail.com Hans, Thank you so much for your response. We will be moving everything to VM soon. Cheers, Nick. On Tue, Nov 8, 2011 at 6:11 PM, Hans Witvliet aster...@a-domani.nl wrote: On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote: That sucks! What about KVM or XEN? Nick. No problems here with XEN. (Perhaps i should mention, that i use paravirtualsisation to get the best performance. Distro: mix of SLES11sp1 /open_11.4) hw -- _ -- 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 -- _ -- 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] Permanent sip and agi debug on?
On 11/09/2011 04:22 AM, David Cunningham wrote: Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. In recent versions of Asterisk, you can put CLI commands into cli.conf and they will be run automatically when Asterisk starts. There are even examples of doing this for 'sip set debug' in cli.conf.sample :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] State of Asterisk+Virtualization+Timing
Hahah... I was waiting on the sideline for this question. Nick. On Wed, Nov 9, 2011 at 8:10 AM, Anton Kvashenkin anton.juga...@gmail.com wrote: Is anybody using pci-passthrough? 2011/11/9 Nick Khamis sym...@gmail.com Hans, Thank you so much for your response. We will be moving everything to VM soon. Cheers, Nick. On Tue, Nov 8, 2011 at 6:11 PM, Hans Witvliet aster...@a-domani.nl wrote: On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote: That sucks! What about KVM or XEN? Nick. No problems here with XEN. (Perhaps i should mention, that i use paravirtualsisation to get the best performance. Distro: mix of SLES11sp1 /open_11.4) hw -- _ -- 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 -- _ -- 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] [OT] Re: Licensing question.
On 11/08/2011 07:54 PM, Raj Mathur (राज माथुर) wrote: On Wednesday 09 Nov 2011, Kevin P. Fleming wrote: [snip] * The GPLv2 places no restrictions on what you can 'write', it only places restrictions on your distribution of things that you write that could be considered 'derivative works' of a GPLv2-covered work (in this case, Asterisk). If you write something that could be considered a derivative work, and you wish to distribute it, then the GPLv2 obligates you to distribute that work under the GPLv2 or a compatible license. Minor nitpick: a derivative of a GPLv2 work can only be released under the GPLv2, or a licence so similar to GPLv2 as to be indistinguishable from it. You cannot distribute a GPLv2 derived work under, e.g. a BSD or Artistic licence. This is strictly true, but of course any decisions to allow or disallow distribution of a derived work are made by the party(ies) who distributed the original work under the GPLv2; if they choose to allow distribution under more permissive licenses, they can certainly do so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] Licensing question.
On 11/09/2011 04:37 AM, A J Stiles wrote: On Tuesday 08 November 2011, Yaroslav Panych wrote: Greetings I have found next paragraph in Licence file(source root) Digium, Inc. (formerly Linux Support Services) holds copyright and/or sufficient licenses to all components of the Asterisk package, and therefore can grant, at its sole discretion, the ability for companies, individuals, or organizations to create proprietary or Open Source (even if not GPL) modules which may be dynamically linked at runtime with the portions of Asterisk which fall under our copyright/license umbrella, or are distributed under more flexible licenses than GPL. What does it mean? Does it mean I can write non-GPL modules(BSD, MIT, etc)? Can I build my modules in common asterisk source tree(i.e. using LOCAL_MOD_SUBDIRS=my_mod_subdirs_list make ) or must use separate tree? If so, then since Asterisk core does not accepts anything except AST_MODULE_INFO(ASTERISK_GPL_KEY, ) what I should do here? If you write modules that need to be compiled against the Asterisk Source Code, then the resulting compiled binaries are by definition derivative works of Asterisk. The GPL already gives you permission to release those modules under the GPL. And Fair Dealing / Fair Use provisions of copyright law mean you need no explicit permission to make use of those modules yourself for their rightful purpose. You require a special, separate licence from Asterisk to distribute compiled binaries which are derived works of Asterisk under anything but the GPL. Other people can, also under Fair Dealing provisions, compile their own legitimately-acquired copy of your module Source Code against their own legitimately-acquired copy of the Asterisk Source Code; but what they end up with may well be unredistributable. What you *can't* do is distribute your modules *as pre-compiled binaries* under any licence beside the GPL -- if they are distributed under any other licence, they *must* be compiled on-site by the end user. This is not true. Distribution in source or binary form makes no difference; if you produce an Asterisk module (that falls under the 'derivative work' classification), whether you distribute it in source or binary form you must distribute it under the terms of the GPLv2 unless you have been granted explicit permission to do otherwise. Of course, as I said in my original reply, anyone who has plans to distribute Asterisk-derived works and wishes to do us under any license other than the GPLv2 would be wise to consult legal counsel in their area to learn how the license affects their plans. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] Licensing question.
I shall contact when(and if) decision will be made. But such decision cannot be made basing only on this paragraph, because it does not describes anything. There are no description of licensing procedure, nor pricing, nor liability, rights or freedoms(at least in general approximation) of sides. So I'm here asking and asking again. In any case, even usage of GPL-ed copy of Asterisk(or any other software) is illegal in my country. regards, Yaroslav. -- _ -- 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] Licensing question.
On 11/09/2011 07:59 AM, Yaroslav Panych wrote: I shall contact when(and if) decision will be made. But such decision cannot be made basing only on this paragraph, because it does not describes anything. There are no description of licensing procedure, nor pricing, nor liability, rights or freedoms(at least in general approximation) of sides. So I'm here asking and asking again. In any case, even usage of GPL-ed copy of Asterisk(or any other software) is illegal in my country. Why would you expect a single paragraph in the Asterisk source tree to have a complete description of Digium's commercial licensing options? It wouldn't even make sense for that to be in the source tree, disregarding that commercial license terms are typically negotiated with each customer based on their unique situation. Your last question doesn't make any sense; you are asking us if usage of GPLv2-licensed software is illegal in your country? We don't even know what country you live in, and even if we did, the answer to that question is something you need to obtain from people who clearly understand your country's laws. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] Licensing question.
On Wednesday 09 Nov 2011, Yaroslav Panych wrote: I shall contact when(and if) decision will be made. But such decision cannot be made basing only on this paragraph, because it does not describes anything. There are no description of licensing procedure, nor pricing, nor liability, rights or freedoms(at least in general approximation) of sides. So I'm here asking and asking again. In any case, even usage of GPL-ed copy of Asterisk(or any other software) is illegal in my country. If I understand your situation correctly, the solution is very simple and two-fold: 1. You want to develop and distribute FOSS (Free and Open-Source Software) extensions for Asterisk. These extensions may be modules or enhancements. In this situation, just go ahead and develop the extensions and distribute them under GPLv2. 2. You want to develop and distribute extensions for Asterisk but don't want to release their source. In this situation, get a source code licence from Digium (or whoever has Asterisk source copyright) under a non-GPLv2 licence. Develop and release your extensions under a proprietary licence. You have no obligation to release their source, since your extensions are derived from a non-GPL licence. Of course, Digium (or whoever) will impose conditions on your use of the source they license to you, but that is between you and Digium (or whoever) and doesn't concern anyone else. This is obviously very generalised and doesn't cover all cases, but it does cover the basic dual-licensing policy. I am not a lawyer. This is not legal advice. In fact, as others have also stated, it is highly recommended you consult with a lawyer who is well-versed in licensing and specially FOSS licensing. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Licensing question.
On Wednesday 09 November 2011, Kevin P. Fleming wrote: On 11/09/2011 04:37 AM, I wrote: What you *can't* do is distribute your modules *as pre-compiled binaries* under any licence beside the GPL -- if they are distributed under any other licence, they *must* be compiled on-site by the end user. This is not true. Distribution in source or binary form makes no difference; if you produce an Asterisk module (that falls under the 'derivative work' classification), whether you distribute it in source or binary form you must distribute it under the terms of the GPLv2 unless you have been granted explicit permission to do otherwise. But so long as you were careful not to copy any of the code you are going to link against into your Source Code (and why would you, if you were linking against it?), it only *becomes* a derivative work *after* it has been compiled. The Source Code for your module is your own independent work (even although it may well be useless without Asterisk) and therefore subject to your own choice of licence. Once it's compiled, the binary contains Asterisk code and so *is* a derivative work. It now falls under a combined licence, and thereafter you can *only* distribute it in accordance with the terms of *both* licences. This is where you risk running afoul of clause 6, no additional restrictions. But as long as you acquired the Asterisk Source Code legitimately, the GPL does not restrict you *using* it (that's a statutory right) -- only *distributing* it (which is a reserved right). Of course, as I said in my original reply, anyone who has plans to distribute Asterisk-derived works and wishes to do us under any license other than the GPLv2 would be wise to consult legal counsel in their area to learn how the license affects their plans. Agreed. If everybody just used the GPL, there wouldn't be any problems with licensing. -- AJS Answers come *after* questions. -- _ -- 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] Licensing question.
But so long as you were careful not to copy any of the code you are going to link against into your Source Code (and why would you, if you were linking against it?), it only *becomes* a derivative work *after* it has been compiled. That's not necessarily true because if you have a work that cannot be used independently (e.g. a plug-in), there are numerous court precedents that say that it indeed is a derived work. This area of the law is very complex and people should really consult an attorney experienced in this area if they care about such things. -- _ -- 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] CallerID inconsistently presented through ISDN/cellular networks
As promised, here is a follow up on my quest to get CallerID correctly presented when forwarding calls to cellphones. Here is a reminder of the issue at hand: Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension) which forwards to Cory (GSM handset) What I would like to get is to see Alice's number (not Bob's number) presented to Cory. Sometimes, I get Alice's number, sometimes, I get Bob's number (new findings from last sunday trials). And of course, if Daniel or Eric would call Bob, the CallerID number presented to Cory would either be Daniel's number, Eric's number or Bob's number depending on a root cause I'm looking after for several days now. To check if CallerID is filtered or controlled by Telco, I originated calls from Asterisk using hand crafted caller ids: any CallerID was correctly presented. So I originally thought the root cause I'm after is a telco equipment switching ANI and CID. But a close look at some last trials output makes me asking for opinions from this list readers. Here follows, the anonymized (and hand indented) output of command PRI debug command. I focused on the end of call setup dialog. For the successfully presented call, the output is: [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [6c 0b 21 83 37 38 36 XX XX XX XX XX XX] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Presentation: Presentation allowed of network provided number (3) '78649' ] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [70 0b 80 30 36 37 31 XX XX XX XX XX XX] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '067100' ] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX XX] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Redirecting Number (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Ext: 0 Presentation: Presentation permitted, user number passed network screening (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Ext: 1 Reason: Forwarded unconditionally (15) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [a1] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Sending Complete (len= 1) For the unsuccessfully presented call, the output is: [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: [6c 0b 21 83 36 37 38 XX XX XX XX XX XX] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Presentation: Presentation allowed of network provided number (3) '67854' ] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: [70 0b 80 30 36 37 31 XX XX XX XX XX XX] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '067100' ] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: [a1] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Sending Complete (len= 1) Am I correctly interpreting when saying that in the successful call, Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX XX] message which is not otherwise sent ? What can explains this difference ? Is this something I can (should) control ? For reference: dahdi show version DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC pri show version libpri version: 1.4.10.2 Improved support for manipulation of redirecting number is available with the REDIRECTING dialplan function in Asterisk v1.8.x and libpri v1.4.12. Prior to Asterisk v1.8.x you only have CALLERID(RDNIS). https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Richard Hi Richard, 1. Could you elaborate a bit ? Do you imply that the lines bellow were present (or missing) because I did somewhere set CALLERID(RDNIS) and that I should use them ? [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Redirecting Number (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Ext: 0 Presentation: Presentation permitted, user number passed network screening (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Ext: 1 Reason: Forwarded unconditionally (15) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ] No. I was trying to say
[asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins,
so in the example below, for example, how can I get the current user count
in ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663)
exten = 2663,1,Answer
same = n,Wait(1)
same = n,Authenticate(143382)
;Record conference callscount: ${count} --)
same = n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
--
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
2011/11/9 Richard Mudgett rmudg...@digium.com As promised, here is a follow up on my quest to get CallerID correctly presented when forwarding calls to cellphones. Here is a reminder of the issue at hand: Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension) which forwards to Cory (GSM handset) What I would like to get is to see Alice's number (not Bob's number) presented to Cory. Sometimes, I get Alice's number, sometimes, I get Bob's number (new findings from last sunday trials). And of course, if Daniel or Eric would call Bob, the CallerID number presented to Cory would either be Daniel's number, Eric's number or Bob's number depending on a root cause I'm looking after for several days now. To check if CallerID is filtered or controlled by Telco, I originated calls from Asterisk using hand crafted caller ids: any CallerID was correctly presented. So I originally thought the root cause I'm after is a telco equipment switching ANI and CID. But a close look at some last trials output makes me asking for opinions from this list readers. Here follows, the anonymized (and hand indented) output of command PRI debug command. I focused on the end of call setup dialog. For the successfully presented call, the output is: [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [6c 0b 21 83 37 38 36 XX XX XX XX XX XX] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Presentation: Presentation allowed of network provided number (3) '78649' ] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [70 0b 80 30 36 37 31 XX XX XX XX XX XX] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '067100' ] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX XX] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Redirecting Number (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Ext: 0 Presentation: Presentation permitted, user number passed network screening (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Ext: 1 Reason: Forwarded unconditionally (15) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: [a1] [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Sending Complete (len= 1) For the unsuccessfully presented call, the output is: [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: [6c 0b 21 83 36 37 38 XX XX XX XX XX XX] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Presentation: Presentation allowed of network provided number (3) '67854' ] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: [70 0b 80 30 36 37 31 XX XX XX XX XX XX] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '067100' ] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: [a1] [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c: Sending Complete (len= 1) Am I correctly interpreting when saying that in the successful call, Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX XX] message which is not otherwise sent ? What can explains this difference ? Is this something I can (should) control ? For reference: dahdi show version DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC pri show version libpri version: 1.4.10.2 Improved support for manipulation of redirecting number is available with the REDIRECTING dialplan function in Asterisk v1.8.x and libpri v1.4.12. Prior to Asterisk v1.8.x you only have CALLERID(RDNIS). https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Richard Hi Richard, 1. Could you elaborate a bit ? Do you imply that the lines bellow were present (or missing) because I did somewhere set CALLERID(RDNIS) and that I should use them ? [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Redirecting Number (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: Ext: 0 Presentation: Presentation permitted, user number passed network screening (1) [Nov
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
2. As I feel specically new to this RDNIS concept, how should I set CALLERID(RDNIS), before or after Answer() statement ? It does not matter in this case. Asterisk v1.6.1 will keep both legs of the call anyway. If you ultimately want to get the call entirely off of your Asterisk server, you will need Asterisk v1.6.2 or later. You would also need libpri 1.4.12 to do this with ETSI(EuroISDN). You would then use the DAHDISendCallreroutingFacility application *before* you answer the call to forward/deflect the incoming call back to the network. Richard That's definitively worth to try. I can't think of any use case but does this DAHDISendCallreroutingFacility generates AMI events, for curiosity's sake ? No. The application just asks libpri to send a FACILITY message to the network. Other AMI events are generated as a result of the redirected call clearing. Richard -- _ -- 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] ConfBridge 1.6.20 user count
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference
identifiers
(0 or 1 for 'locked') and for invalid conference identifiers.
[Syntax]
CONFBRIDGE_INFO(type,conf)
[Arguments]
type
Type can be 'parties', 'admins', 'marked', or 'locked'.
conf
Conf refers to the name of the conference being referenced.
Guess the developers of confbridge didn't want to duplicate the meetme_count
function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins,
so in the example below, for example, how can I get the current user count
in ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663)
exten = 2663,1,Answer
same = n,Wait(1)
same = n,Authenticate(143382)
;Record conference callscount: ${count} --)
same = n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
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Re: [asterisk-users] ConfBridge 1.6.20 user count
Unfortunately, that function doesn't seem to be in 1.6.20, which
Asterisk version are you using?
*CLI core show function CONFBRIDGE_INFO
No function by that name registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote:
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference identifiers
(0 or 1 for 'locked') and for invalid conference identifiers.
[Syntax]
CONFBRIDGE_INFO(type,conf)
[Arguments]
type
Type can be 'parties', 'admins', 'marked', or 'locked'.
conf
Conf refers to the name of the conference being referenced.
Guess the developers of confbridge didn’t want to duplicate the meetme_count
function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins, so
in the example below, for example, how can I get the current user count in
ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663)
exten = 2663,1,Answer
same = n,Wait(1)
same = n,Authenticate(143382)
;Record conference callscount: ${count} --)
same = n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
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Re: [asterisk-users] ConfBridge 1.6.20 user count
10.0.beta2. Have you tried confbridge(xxx,c)? This joins and announces
count, but I don't know if it returns a variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ConfBridge 1.6.20 user count
Unfortunately, that function doesn't seem to be in 1.6.20, which Asterisk
version are you using?
*CLI core show function CONFBRIDGE_INFO No function by that name
registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote:
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference
identifiers
(0 or 1 for 'locked') and for invalid conference identifiers.
[Syntax]
CONFBRIDGE_INFO(type,conf)
[Arguments]
type
Type can be 'parties', 'admins', 'marked', or 'locked'.
conf
Conf refers to the name of the conference being referenced.
Guess the developers of confbridge didnt want to duplicate the
meetme_count function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins,
so in the example below, for example, how can I get the current user count
in ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663) exten = 2663,1,Answer
same = n,Wait(1) same = n,Authenticate(143382)
;Record conference callscount: ${count} --) same =
n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
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Re: [asterisk-users] ConfBridge 1.6.20 user count
confbridge(xxx,c) is a blocking call, so you can't get status back
until that command completes. Time to upgrade to 10.0.beta2 I
guess...
On Wed, Nov 9, 2011 at 12:47 PM, Danny Nicholas da...@debsinc.com wrote:
10.0.beta2. Have you tried confbridge(xxx,c)? This joins and announces
count, but I don't know if it returns a variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ConfBridge 1.6.20 user count
Unfortunately, that function doesn't seem to be in 1.6.20, which Asterisk
version are you using?
*CLI core show function CONFBRIDGE_INFO No function by that name
registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote:
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference
identifiers
(0 or 1 for 'locked') and for invalid conference identifiers.
[Syntax]
CONFBRIDGE_INFO(type,conf)
[Arguments]
type
Type can be 'parties', 'admins', 'marked', or 'locked'.
conf
Conf refers to the name of the conference being referenced.
Guess the developers of confbridge didn’t want to duplicate the
meetme_count function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins,
so in the example below, for example, how can I get the current user count
in ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663) exten = 2663,1,Answer
same = n,Wait(1) same = n,Authenticate(143382)
;Record conference callscount: ${count} --) same =
n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
--
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Re: [asterisk-users] ConfBridge 1.6.20 user count
What about a local call to confbridge(xxx,c)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 12:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ConfBridge 1.6.20 user count
confbridge(xxx,c) is a blocking call, so you can't get status back until
that command completes. Time to upgrade to 10.0.beta2 I guess...
On Wed, Nov 9, 2011 at 12:47 PM, Danny Nicholas da...@debsinc.com wrote:
10.0.beta2. Have you tried confbridge(xxx,c)? This joins and
announces count, but I don't know if it returns a variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
users
Sent: Wednesday, November 09, 2011 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ConfBridge 1.6.20 user count
Unfortunately, that function doesn't seem to be in 1.6.20, which
Asterisk version are you using?
*CLI core show function CONFBRIDGE_INFO No function by that name
registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote:
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference
identifiers
(0 or 1 for 'locked') and for invalid conference identifiers.
[Syntax]
CONFBRIDGE_INFO(type,conf)
[Arguments]
type
Type can be 'parties', 'admins', 'marked', or 'locked'.
conf
Conf refers to the name of the conference being referenced.
Guess the developers of confbridge didnt want to duplicate the
meetme_count function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user
joins, so in the example below, for example, how can I get the current
user count in ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663) exten = 2663,1,Answer
same = n,Wait(1) same = n,Authenticate(143382)
;Record conference callscount: ${count} --) same =
n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},b
m))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
--
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Re: [asterisk-users] Permanent sip and agi debug on?
Kevin, Thank you very much! On 10 November 2011 00:15, Kevin P. Fleming kpflem...@digium.com wrote: On 11/09/2011 04:22 AM, David Cunningham wrote: Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. In recent versions of Asterisk, you can put CLI commands into cli.conf and they will be run automatically when Asterisk starts. There are even examples of doing this for 'sip set debug' in cli.conf.sample :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Permanent sip and agi debug on?
If you have an ancient version of Asterisk you want to stick with, you can do this with asterisk -rx sip set debug on and asterisk -rx agi set debug on in your safe_asterisk script. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Wednesday, November 09, 2011 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Permanent sip and agi debug on? Kevin, Thank you very much! On 10 November 2011 00:15, Kevin P. Fleming kpflem...@digium.com wrote: On 11/09/2011 04:22 AM, David Cunningham wrote: Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. In recent versions of Asterisk, you can put CLI commands into cli.conf and they will be run automatically when Asterisk starts. There are even examples of doing this for 'sip set debug' in cli.conf.sample :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] State of Asterisk+Virtualization+Timing
On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote: Is anybody using pci-passthrough? Yes, though quite a while ago. About three years ago, i used pci-passthrough to give a dom-U access to a localy mounted smartcard. But i have a vague feeling that you are up to something else... I know that forwarding has been done for ethernet and even VGA-cards, the mere idea of forwarding a analogue or PRI card is quite something else: Timing is here essential.. hw -- _ -- 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] State of Asterisk+Virtualization+Timing
Smart card? I think we should be leaning more towards the network devices? Cheers, Nick. On Wed, Nov 9, 2011 at 5:23 PM, Hans Witvliet aster...@a-domani.nl wrote: On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote: Is anybody using pci-passthrough? Yes, though quite a while ago. About three years ago, i used pci-passthrough to give a dom-U access to a localy mounted smartcard. But i have a vague feeling that you are up to something else... I know that forwarding has been done for ethernet and even VGA-cards, the mere idea of forwarding a analogue or PRI card is quite something else: Timing is here essential.. hw -- _ -- 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
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[asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks
Hi All, I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in routing calls to upstream carrier via SIP trunks out. I spent a lot of time in the lab testing 1.8 which included heavily testing DTMF with no issues that came up. It all just seemed to work fine. But then again you can't reproduce every real work scenario in the lab. I'm using rfc2833 inbound and outbound for the new 1.8 call servers. Here is a quick diagram of what is working and what is not: Not working: Customer IP PBXsip trunk rfc2833ast 1.4 rfc2833sip trunkcall server ast 1.8 rfc2833sip trunkupstream carrier Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833sip trunk call server ast 1.8 rfc2833sip trunkupstream carrier I can see DTMF RTP events pass through call server to carrier but no response, nothing, nada, zip. Working: Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server ast 1.8 rfc2833sip trunkupstream carrier Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server ast 1.2 rfc2833sip trunkupstream carrier Customer IP PBXsip trunk rfc2833ast 1.4 rfc2833sip trunk call server ast 1.2 rfc2833sip trunkupstream carrier Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833 call server sip trunkast 1.2sip trunkupstream carrier I can see DTMF RTP events pass through to carrier, RTP stream looks the same as the 1.8 server with reliable responses. On both the 1.4 and 1.8 ast servers, these sip.conf parameters are active on peer and global settings: relaxdtmf=yes rfc2833compensate=yes dtmfmode=rfc2833 Now it quickly appears like a problem between the customer PBX and Customer PRI with the SIP trunks to the ast 1.4 servers but it all worked fine before with the 1.2 call servers. After the upgrade of the call servers to 1.8 DTMF is not recognized by the carrier on calls from the customer IP PBX or PRI but is fine with the SIP phones directly registered to the ast 1.4 servers. I found the bug issues with the SRCC change/update issues with DTMF events. It looks like 1.8.6.0 implemented the 'update' and as I read it, should have fixed the issue with the changing SRCC effecting DTMF. But this may not be the case. Specifically, how would I debug RTP/DTMF on the new ast 1.8 server and see if the SRCC is changing between my scenarios described above. Am I on the right track or is there something else I should be looking at? Thanks. JR -- _ -- 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] DTMF issue with 1.8.6.0 and SIP Trunks
I had similar problems with 1.8.6 and polycom phones intermittently having DTMF issues. I updated to 1.8.7 and things cleared up. I went through the release notes at the time, but don't recall which commit made me decide to give it a try. Rgds, Jared On Wed, Nov 9, 2011 at 7:03 PM, JR Richardson jmr.richard...@gmail.comwrote: Hi All, ** ** I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in routing calls to upstream carrier via SIP trunks out. I spent a lot of time in the lab testing 1.8 which included heavily testing DTMF with no issues that came up. It all just seemed to work fine. But then again you can’t reproduce every real work scenario in the lab. ** ** I’m using rfc2833 inbound and outbound for the new 1.8 call servers. Here is a quick diagram of what is working and what is not: ** ** Not working: Customer IP PBXsip trunk rfc2833ast 1.4 rfc2833sip trunkcall server ast 1.8 rfc2833sip trunkupstream carrier ** ** Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833sip trunk call server ast 1.8 rfc2833sip trunkupstream carrier ** ** I can see DTMF RTP events pass through call server to carrier but no response, nothing, nada, zip. ** ** Working: Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server ast 1.8 rfc2833sip trunkupstream carrier ** ** Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server ast 1.2 rfc2833sip trunkupstream carrier ** ** Customer IP PBXsip trunk rfc2833ast 1.4 rfc2833sip trunk call server ast 1.2 rfc2833sip trunkupstream carrier ** ** Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833 call server sip trunkast 1.2sip trunkupstream carrier ** ** I can see DTMF RTP events pass through to carrier, RTP stream looks the same as the 1.8 server with reliable responses. ** ** On both the 1.4 and 1.8 ast servers, these sip.conf parameters are active on peer and global settings: relaxdtmf=yes rfc2833compensate=yes dtmfmode=rfc2833 ** ** Now it quickly appears like a problem between the customer PBX and Customer PRI with the SIP trunks to the ast 1.4 servers but it all worked fine before with the 1.2 call servers. After the upgrade of the call servers to 1.8 DTMF is not recognized by the carrier on calls from the customer IP PBX or PRI but is fine with the SIP phones directly registered to the ast 1.4 servers. ** ** I found the bug issues with the SRCC change/update issues with DTMF events. It looks like 1.8.6.0 implemented the ‘update’ and as I read it, should have fixed the issue with the changing SRCC effecting DTMF. But this may not be the case. ** ** Specifically, how would I debug RTP/DTMF on the new ast 1.8 server and see if the SRCC is changing between my scenarios described above. Am I on the right track or is there something else I should be looking at? ** ** Thanks. JR -- _ -- 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
[asterisk-users] IAX2 availability testing
Hi folks, What methods are available for testing IAX2 service availability? I know about iax2 show peers and iax2 show registry, but I'd like some alternatives. Tcpdump shows a little more about what's going on, but a handy test using nmap doesn't seem to work anymore (see http://shearer.org/UDP_Reachability_Testing). Any suggestions would be appreciated. Cheers, Jaap PS -- My systems run Debian squeeze with Asterisk 1.6.2.9. -- _ -- 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] 9. any live queue monitor recommendation? (Jean Chassoul) chass...@gmail.com
I would recommend you Monast (monitor asterisk), it's stable and gives alot of information. http://sourceforge.net/projects/monast/ WBR A.Rymkus 04.11.2011 13:34, Anthony Laudini ?: Hi Jean, I suggest Queuemetrics. There are many out there but this one is good for monitoring and reporting. I know there's a free version you can try. All the best Anthony -- _ -- 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
