[asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause
code) commands, if the call is not answered by peer01 for any reason,
the actual cause code returned to the calling party is a 503, no matter
what the ${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always
a 503.
Has anyone else noticed this? I went through the issue tracker but I
couldn't find any relevant bug posted in the past. I am certain that in
previous versions I could set the reply message to the desired value, so
I wonder if this is a bug in this particular version (1.4.33.1).
--
Best regards,
Vlasis Hatzistavrou.
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It
is that the Hangup(cause) command seems to ignore its argument and
just sends a 503 cause to the caller for all unanswered calls no matter
what...
Hangup(cause) was working as expected in previous versions and I
wonder if something was broken along the way that went by unnoticed. I
am just asking in the list in case I am missing something too obvious
before posting a bug.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 4:22 μμ, Jim Dickenson wrote:
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
The h extension is executed after the remote end peer01 rejects the call
with a 408. I verified it by altering the dialplan as:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,NoOp(Hangup cause is: ${HANGUPCAUSE})
exten = h,n,Hangup(${HANGUPCAUSE})
and I saw in the Asterisk CLI that the correct hangupcause is shown.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 5:01 μμ, Jim Dickenson wrote:
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
Hello Steve, On 15/4/2011 5:07 μμ, Steve Davies wrote: Strictly speaking you can only Hangup (BYE) an answered and fully established call. In SIP terms, a hangup that occurs before an answer is a CANCEL, and I believe a CANCEL is always represented by a 503 code in chan_sip. Regards, Steve I see what you mean, but it is the called end (peer01) that rejects the call with a 408 message, it is not the originator that is canceling the call. The call flow is this: Caller-Asterisk-Peer01 and Asterisk receives a STATUS 408 message from Peer01 instead of an answer. Asterisk then sends a STATUS 503 to the Caller, instead of sending a STATUS 408. The question is how to copy the correct cause code from the terminating end to the originating end. I tried setting Hangup(1) to send a 404 to the called, a Hangup(17) to send a 486 to the caller and pretty much any other value in the Hangup() but Asterisk will keep on sending a 503. I don't believe that my memory fails me, I'm pretty sure I could set a desirable cause in the Hangup() command in previous versions... -- Best regards, Vlasis Hatzistavrou. -- _ -- 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] Dial option 'r' not working anymore?
Hello,
I have used the Dial option 'r' before in older Asterisk versions and it
behaved as expected, that is, Asterisk would always give ringback audio
before the call was answered no matter what.
It has been a while that I have used version 1.4.33.1 without any the
'r' option. Now that I had to use it for a while, I noticed that 'r'
would not give ANY audio until the call was answered.
I looked up the documentation of app Dial, but nothing new was
mentioned, compared to the known 'r' behavior.
I also Googled it, looked through the mailing list, but I couldn't find
anything to help me. In fact, I noticed that there was a lot of confused
questions and even confused/confusing answers about the behavior of 'r'.
The extension that I use is pretty simple:
exten = _X.,1,Dial(SIP/${numb...@x.y.z.w,,r,)
Does anyone know if the behavior of 'r' has changed but was not
documented? If yes, then how does one inject ringback audio before the
call is answered on the called end?
--
Best regards,
Vlasis Hatzistavrou.
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Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Olivier wrote: Hi, I've been playing with T.38. I observed that mostly but not always, it's the calling endpoint that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the standardized or most common, way to start a T.38 session ? Shall it come from callee or from caller ? Regards Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. ___ -- 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] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Vlasis Hatzistavrou (KTI) wrote: Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Steve Underwood wrote: Hey, why bother looking at a spec when its so much more fun to make it up as we go along? ... Regards, Steve I don't think there is a need to be ironic here... I wrote AFAIK which we all know means as far as I know, so why the bashing? ___ -- 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] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Olivier wrote: 2009/3/17 Vlasis Hatzistavrou (KTI) vh...@kinetix.gr mailto:vh...@kinetix.gr Vlasis Hatzistavrou (KTI) wrote: Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Steve Underwood wrote: Hey, why bother looking at a spec when its so much more fun to make it up as we go along? ... Regards, Steve I don't think there is a need to be ironic here... I wrote AFAIK which we all know means as far as I know, so why the bashing? Vlasis, I don't think Steve's irony where targeted to you but to those which are supposed to read specs (ATA vendors) ... Hello Olivier, Well, since Steve's comment followed right after my reply, it seemed like the comment was very much targeted at me... The comment can be taken both ways I guess... Regards, Vlasis. ___ -- 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] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Steve Underwood wrote: Oh, it was meant for him. In the time it took him to write his wrong e-mail he could have gone to the ITU web site, downloaded a free copy of the T.38 spec, looked up the annex where it described the negotiation process, and found a clear statement of what is supposed to happen. Of course, that wouldn't tell him the real world issues, like the fact half the T.38 implementations out there don't follow the spec., but it would have been a valuable start. It would also keep the noise level on this list down. What a lot of people don't allow for when writing garbage is it stays on the internet for years, and eventually becomes reference material. :-\ Regards, Steve Does AFAIK mean anything at all to you? I never implied that I am the ultimate authority on fax. It has been many years since I read T38 or any other fax specs and apparently I don't remember them to the letter (hence the AFAIK in my sentence). Reference material? Really? My reply on a mailing list can hardly be mistaken for an ITU spec. The fact that my email will remain on the internet for years cannot justify your obnoxious behavior either, unless you honestly believe that my post will misguide the future generations of VoIP implementors for years to come... In other words, if you really wanted to correct my mistake you could have just said that I was wrong. I would even have thanked you for pointing out my error. In such a scenario you would have really contributed against the noise on this list. But unfortunately, all you did was come out as just another wise-guy who desperately needs to get off his high horse. Cheers, Vlasis. ___ -- 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] Country numbering plan resources
For informational purposes many people find ITU's web site useful, although not always as detailed as one would probably want: http://www.itu.int/itu-t/inr/nnp/index.html It even has event dates of official numbering plan changes. Best regards, Vlasis Hatzistavrou Kinetix Tele.com International Inc. 306 Victoria House, Victoria, Mahe, Seychelles Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: vh...@kinetixtele.com http://www.kinetixtele.com Postal address: Monastiriou 9 Enotikon 54627 Thessaloniki Greece ___ -- 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] CDR Design
Hello Andy, But, wouldn't it be better if you could ignore the CDR's completely and use an event based system? This would give you ALL the information you need. All that remains is to filter out the un-required bits. I'd disagree. In some cases a event based system would be the best solution, but in systems with high call volumes, scanning through events looking for the proper billing information and parsing them would be a hard job compared to CDRs. In most cases where there are no transfers, calls on hold etc, but only basic dial-in dial-out operations, using events instead of CDRs would probably be an overkill. Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). I'm not sure I understand what you mean exactly. If you have in mind cases of transfers, calls on hold etc and you refer to Asterisk's CDRs at this point in time, then indeed, Asterisk's CDRs are not reliable in many cases. However, CDRs in general on other platforms tend to be very reliable and useful for billing. My opinion is to transform Asterisk's CDR capabilities to something more carrier-grade in mind, configurable by the user. Best regards, Vlasis Hatzistavrou. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: 05 December 2008 09:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR Design On Fri, Dec 5, 2008 at 8:26 AM, Andrew Thomas [EMAIL PROTECTED] wrote: In summary: Leave CDR exactly as it is and create a new CEL (Call Event Logging) module (optional in modules.conf) that puts out (and does not accept) call event information (ie. a one-way fire-and-forget output from Asterisk). Hi Andrew and Others, This thread is actually part of a discussion that has been going on for over a year. The links below provide the background to the whole thing. http://www.asterisk.org/node/48358 http://bugs.digium.com/view.php?id=11849 http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.htm l Up until recently the approach was to try and fix the specific bugs with transfer CDRs as a typical bug. There is now a realisation that that is a lot trickier than inially thought so it's been decided to try and come up with a good design for the Asterisk CDR sub-system. Regards, Greyman. ___ -- 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
Re: [asterisk-users] CDR Design
Hello, Andrew Thomas wrote: I'd disagree. In some cases a event based system would be the best solution, but in systems with high call volumes, scanning through events looking for the proper billing information and parsing them would be a hard job compared to CDRs. That's just it - you wouldn't be 'scanning' any CDR's - you'd be given Events. Your 3rd party app could then do anything it wanted to with them. Events are real time - not historic (like CDR's). Events are presented as they happen (hold, ring, etc) - CDR's are usually presented AFTER the call has finished so you miss things like hold-times etc. Indeed, if you refer to real time events then this is the way to go. However, many people (including our company) use CDRs as a fall back in case we don't have real-time billing data available. We use real time information for prepaid customers and stats, but we also crosscheck this data with CDRs periodically. I agree that both approaches would be useful in many different scenarios for different users. It would be ideal if both approaches could be implemented. Best regards, Vlasis Hatzistavrou. ___ -- 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] CDR Design
Andrew Thomas wrote: Quote : I couldn't disagree more. The CDRs is the MOST reliable source for billing purposes ...at the moment. Have you read about Greyman's transfer problem? If you are billing customers purely on the CDR output from Asterisk - then good luck to you :). This is exactly our point in this discussion. :):) We can't bill relying on Asterisk's CDRs at this moment, this is why we use a third party SBC to do real time billing stats, as well as collect CDRs from the SBC off-line for cross-checking with the live data. And this is why we support the opinion that Asterisk's CDRs should be expanded. Best regards, Vlasis Hatzistavrou. ___ -- 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] OT - Is sourceforge OpenH323 active ?
As I recall, when openh323.org because obsolete people could download the PWLib OpenH323 libraries from http://www.voxgratia.org/ Openh323 has moved to become OPAL (supporting SIP, H323 and IAX) and can be downloaded from http://www.opalvoip.org H323Plus is also a continuation of OpenH323 supporting only H323. If you need to download OpenH323 and PWLib version suitable for Asterisk's chan_h323 you can follow the OpenH323 downloads link at the Voxgratia site. I hope this helps. Best regards, Vlasis Hatzistavrou. Kinetix Tele.com International Inc. 306 Victoria House, Victoria, Mahe, Seychelles Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: [EMAIL PROTECTED] http://www.kinetixtele.com Postal address: Monastiriou 9 Enotikon 54627 Thessaloniki Greece Olivier wrote: Hi, A glance at sourceforge.net/projects/openh323 http://sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers ___ -- 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
Re: [asterisk-users] AGI and prepaid billing + Radius
Yes, of course you can. We have used Perl and Authen::Radius in the past to create AGI calling card scripts to do AAA against RADIUS servers. Not only that, but we used it for routing the outgoing calls also in many cases. Best regards, Vlasis Hatzistavrou. bilal ghayyad wrote: Yes it answer and big thanks. I have another question (which might be not related alot to AGI) if u can help me: If Asterisk support Radius, so we can build Prepaid Billing with Radius to communicate via Radius as standard communication method? Regards Bilal --- On Tue, 9/23/08, Benjamin Jacob [EMAIL PROTECTED] wrote: From: Benjamin Jacob [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Date: Tuesday, September 23, 2008, 6:39 AM Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- 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 ___ -- 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 ___ -- 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] chan_h323 requirements
Hello, To compile chan_h323 as is distributed you need to download OpenH323 v1.18.0 and PwLib v1.10.0 from: http://www.voxgratia.org Some months ago I had made a patch to compile the 1.4.x version and the trunk version (which evolved to 1.6.x) with H323+. Sadly, the patch was not included in the 1.6.x version which is being released soon. So, for the time being you need to use the above versions from Voxgratia. Best regards, Vlasis Hatzistavrou. Bruce McAlister wrote: Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- 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
Re: [asterisk-users] chan_h323 requirements
Hello Bruce, Bruce McAlister wrote: Did your patch for building with OpenH323+ make it into the 1.4 edition of Asterisk? No, it didn't as it was considered a new feature and by Digium's policy new features can only be added in the trunk versions. The strange thing is that I added it in trunk version, too, but it didn't make it in the upcoming 1.6 version either. Best regards, Vlasis Hatzistavrou ___ -- 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] DS3 Interface
Patrick wrote: On Tue, 2007-10-09 at 10:14 -0400, Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? I can imagine it be used as a TDM-SIP gateway but if I needed such a box I'd rather go for a Lucent MaxTNT, Lucent APX8000 or a Cisco 5xxx or look at FreeSWITCH which by design seems more suitable for these kind of high performance applications. There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Why wouldn't today's powerful quadcore servers with Gigabit Ethernet interfaces not be able to handle less than 100Mbit/s synchronous traffic? Please enlighten me as I am no expert here. Regards, Patrick Perhaps it could also be used as a pure TDM switch with no VoIP calls involved? Best regards, Vlasis Hatzistavrou. ___ --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] Re: does OOH323 channel support Early Media?
Hello, There was a patch a couple of weeks ago which would fix the early audio issue of OOH323 when the incoming call was in H323. Unfortunately, there is still a problem with outgoing calls in OOH323 because the incoming CONNECT message is not handled properly, at least in the installations that we tried. Best regards, Vlasis. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rafael J. Risco G.V. Sent: Παρασκευή, 1 Σεπτεμβρίου 2006 1:29 πμ To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re: does OOH323 channel support Early Media? hi early media works with incoming sip calls but for h323 I only hear a ringback instead of 'playback' so I think i am close to solve this, I am just need to know how to disable generating ringback tone for incoming h323 calls, any idea? thanks rafael On 8/31/06, Rafael J. Risco G.V. [EMAIL PROTECTED] wrote: Hi I am trying to send some incoming h323 calls to an early media announcement instead of ringback tone : in extension.conf : exten = 201,1,Playback(thank-you-for-calling|noanswer) is it possible? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conflict between S L option in Dial?
Hello, I have a question about whether the L option in the dial command conflicts with the S option? For example, I have the following in my Dial command: SIP/[EMAIL PROTECTED]|60|HL(:3:1)S(120) I see in my CDRs that there are calls lating more than 120 seconds. By reading the description of the dial application, I assumed that since I want the user to talk for 120 seconds I should use: S(120) instead of L(12:3:1) Does anyone know if using L(:3:1) and S(120) conflict each other? Thank you, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] AGI onAnswer function: does it exist?
Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? Best regards, Vlasis Hatzistavrou. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlasis Hatzistavrou Sent: Thursday, February 16, 2006 3:43 PM To: asterisk-users@lists.digium.com Cc: 'Vlasis Hatzistavrou' Subject: [Asterisk-Users] AGI onAnswer function: does it exist? Hello, I am trying to write an AGI in Perl and I need to execute a function upon answer of a call. I know that there is the possibility to use the M() option in the Dial command in order to do what I need, but that would mean that I would have to incorporate the function's work in a different AGI program, and I need to avoid this. So, I would like to know if such an option is available in AGI like an onanswer() function or something equivalent that I can use. Any help would be really appreciated, as I've been searching www.voip-info.org and the Asterisk mailing lists for days now, without any success. Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI onAnswer function: does it exist?
Hello, I am trying to write an AGI in Perl and I need to execute a function upon answer of a call. I know that there is the possibility to use the M() option in the Dial command in order to do what I need, but that would mean that I would have to incorporate the function's work in a different AGI program, and I need to avoid this. So, I would like to know if such an option is available in AGI like an onanswer() function or something equivalent that I can use. Any help would be really appreciated, as I've been searching www.voip-info.org and the Asterisk mailing lists for days now, without any success. Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_bluetooth and Ericcson T68 problem
Hello Enky, We have encountered similar problems with various Ericsson Nokia phones. We couldn't get the channel driver to work 100%. However, we cannot actually tell whether it was our mistake or whether there was a problem with the channel driver. We tried to contact the driver's maintainer/creator but no luck... If you manage to find a solution for this problem we'd also be interested to know about it. Best regards, Vlasis. Enky wrote: Hi, I have read many pages and tried many things, but without any success. I have paired my ERICCSON T68 with the Asterisk PC. The Asterisk version is “Asterisk CVS-v1-0-11/19/05-14:52:52”. The chan_bluetooth is the last release, downloaded from “http://www.crazygreek.co.uk/data/pages/chan_bluetooth/latest.tar.gz”. It is all OK. I can dial from the Asterisk a number. The T68 dials it, but when the called party picks the phone and the call goes connected there is no any audio! Neither from or to the Asterisk. Here are a short logs: This is the initial log, when I start the Asterisk and it connects the T68. It seems OK: ---cut--- Asterisk Ready. *CLI Nov 19 15:15:45 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? Nov 19 15:15:46 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:417 set_cind: Audio Gateway T68 got signal [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- This is when I dial a number. It seems OK too, but no audio when connects: ---cut--- -- Executing Dial(SIP/222-3885, BLT/T68/123|60) in new stack [AG]T68 ATD123; -- Called T68 [AG]T68 OK [AG]T68 +CIEV: 8,1 -- BLT/T68 answered SIP/222-3885 [AG]T68 +CIEV: 2,4 [AG]T68 +CIEV: 2,5 ---cut--- And this is when I interrupt the dialed call: ---cut--- [AG]T68 AT+CHUP == Spawn extension (default, 2002, 1) exited non-zero on 'SIP/222-3885' [AG]T68 OK Nov 19 15:18:06 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2493 rd_close: Device T68 disconnected, scheduled reconnect in 5 seconds: Connection reset by peer (errno 104) Nov 19 15:18:11 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- Please someone to help me :) Thank you in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
Angelito Manansala wrote: yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! Hello, As far as I know Asterisk cannot disentangle RTP from signaling in either SIP or H323 at least until now. I'd also be interested to know if this option is available now in case I've missed something... Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server query
Avi Miller wrote: Hello gurus! I've given up on crappy passive ISDN cards and am heading into the wild world of real, Active Super Dooper Server boards. I have a choice of two Eicon Diva Server cards: Eicon Diva Server 4BRI Eicon Diva Server V-4BRI Hello, We've been using an Eicon Diva Server 4BRI with a RH 9 installation (kernel 2.4.20-8). It works great in both TE and NT mode. I assume that it will work equally great with a 2.6 kernel... Best regard, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Problem with get_data: Caching or ignoring DTMF Tones? (WAS: Re: [Asterisk-Users] Caching DTMF tones for get_data AGI?)
Hello, I'm facing a similar problem, only that in my case, there in no input at all. I use an agi built with Perl and Asterisk::AGI. The $AGI-get_data(...) is executed as if the # was pressed immediately. The strange thing is that this strange behavior happens only if I send a RADIUS accounting packet (using Authen::Radius) to a RADIUS server from the same agi, just before I call the $AGI-get_data. If I comment out the line where I send the RADIUS accounting packet, then $AGI-get_data works fine... Has anyone else dealt with such problems? Best regards, Vlasis Hatzistavrou. Nathan Pralle wrote: I'm using get_data in an AGI script and am having a problem when, after a long time in my IVR, when I ask for a 10-digit phone number, the first few tries are always invalid -- the number it reads back is very strange, almost like the DTMF tones from other answers were being cached and then dumped on the call to get_data. Anyone ever experienced this before? I have to do some major exploring, but nothing comes to mind yet. Nathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_bluetooth and audio problem
Hello,
We had similar problems with chan_bluetooth and various mobile devices.
I suppose that chan_bluetooth is in a very early stage. We tried to
contact the author of the channel with debugging information etc but
without luck...
There is also the chance that the project may be stalled...
Best regards,
Vlasis Hatzistavrou.
José Luis Gómez wrote:
Hello.
I'm having problem with motorola v635 and asterisk. I can make a call
but I can't hear any audio and the other side of the call can hear me
(one way audio).
I'm using usb to bluetooth adaptor (noganet).
I'm using gentoo with kernel 2.6.13-r2, asterisk 1.0.9 and
chan_bluetooth 0.0.1_pre20050212.
What's may be wrong?
I show you my files:
- bluetooth.conf:
[general]
interface = 0
[00:15:A8:A8:19:82]
name= V635
type= HS
channel = 3
autoconnect = yes
# If I put channel 7, the other side of the call can't hear me (no
audio). The audio stay on the phone (I can hear the call on phone).
- hcid.conf
options {
autoinit yes;
security auto;
pairing multi;
pin_helper /usr/bin/bluepin;
}
device {
name Asterisk;
class 0x200404;
iscan enable;
pscan enable;
lm accept;
lp rswitch,hold,sniff,park;
}
- rfcomm.conf
rfcomm0 {
bind yes;
device xx:xx:xx:xx:xx:xx;
channel 7;
comment motoV635;
}
Thanks in advance.
José Luis
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Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)
If anyone is interested I'm (slowly) developing a GPL'd Java applet that works as an IAX softphone. I should have a test version out at the end of the week for a limited number of testers. Tim. http://www.westhawk.co.uk/ Hello Tim, We'd be interested to test the client... Best regards, Vlasis Hatzistavrou Technical Director CEO Kinetix Tele.com Hellas Ltd. Monastiriou 9 Enotikon 546 27 Thessaloniki Greece Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: [EMAIL PROTECTED] http://www.kinetix.gr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP proxies Asterisk ?
Hello, We hve been trying to make Asterisk work with SIP proxies with no success. Is there support for SIP proxies in Asterisk in the latest versions? Best regards, Vlasis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP proxies Asterisk ?
Hi Olle, When we accept calls from a SIP proxy without regitration from either side, but with only an INVITE message, the calls fail. If we set the remote proxy to send us the calls by proxying both RTP signaling, then there is no problem. So, we concluded that Asterisk doesn't like it when signaling and RTP come from different IP addresses. Is there a setting on Asterisk which could allow this? I can provide packet captures if you want. Best regards, Vlasis. Olle E. Johansson wrote: Vlasis Hatzistavrou wrote: Hello, We hve been trying to make Asterisk work with SIP proxies with no success. Is there support for SIP proxies in Asterisk in the latest versions? A lot of people use Asterisk with SIP proxys. What is your problem, give us a bit more information. /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP calls with different signaling and RTP IP addresses
Hello, I use Asterisk CVS-v1-0-12/21/04-11:05:29 and I noticed that when we receive calls from a partner's IP address (who has a static host entry in the sip.conf file) but the RTP comes from a different address than the signaling, our * sends a 403 forbidden message and drops the call. This problem does not llow us to receive calls from SIP proxies. Was this fixed in newer versions of Asterisk? Best regards, Vlasis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internet bandwidth
Hello Michael, Sorry for the late reply. The 17kbps are for the G723.1 at 6.3kbps. The additional overhead which increases the bandwidth usage etc depends on the codec. It's not a fixed overhead in bandwidth for all codecs. You can find a few free codec/bandwidth calculators at: http://www.voipcalculator.com http://www.packetizer.com Best regards, Vlasis. Michael Vogel wrote: Hi! Vlasis Hatzistavrou schrieb: 6.3kbps of G723.1 will become around 17kbps on the IP level without silence suppression because of the additional overhead imposed by protocols like RTP, IP, etc . These 17kbps are they independent from codec? That means a A-LAW with 64kbps has got 64+17=81kbps? BTW: I have seen different descriptions regarding the rate of U-LAW. Is it 64 or 56kbps? If you chose IAX instead of SIP, you will save lots of bandwidth if all (or most) of those 20 calls are directed to the same host. Does IAX save bandwith on single calls as well? Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internet bandwidth
kido noagbodji wrote: Hi Hammoud, It all depends on the codec that you are using. Best case scenario is with G723 codec 6.3Kbps per channel * 20, around 126K without the overhead. But you problably won't be able to use this codec unless you are in passthru mode (license is pretty expensive). Using g729 you will be using 8K so a total of 240K+ total bandwidth (passthru OK but you can purchase the license from digium)... Kido - Original Message - From: chawki hammoud mailto:[EMAIL PROTECTED] To: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Sent: Thursday, November 18, 2004 7:55 AM Subject: [Asterisk-Users] internet bandwidth Hi everybody: How much internet bandwidth and spees is enough to handle twenty simultanous SIP calls. Hello, Some corrections are needed: 6.3kbps of G723.1 will become around 17kbps on the IP level without silence suppression because of the additional overhead imposed by protocols like RTP, IP, etc . If you add the Ethernet (or WAN protocol overhead) this will increase even more (although slightly). Similarly, a voice stream of G729 at 8kbps will become around 24kbps on the IP level, and slightly more on the Ethernet or ppp level (around 25 kbps). So, for 20 channels of 6.3kbps G723.1, you will need around 340 kbps on the IP level without silence suppression. For 20 channels with G729, you will need around 480 kbps. And of course, these calculations apply both ways (upstream downstream). If you chose IAX instead of SIP, you will save lots of bandwidth if all (or most) of those 20 calls are directed to the same host. Best regards, Vlasis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quintum A800 OH323 problem
Hello, I am passing traffic between Asterisk and A800's with OH323 without problems. No calls are disconnected after 20 seconds. Which version of Asterisk, OH323, pwlib and openh323 are you running? Best regards, Vlasis. Ender Erbey wrote: Hi, I experienced an interesting problem when i try to make such a connection BRI -- Asterisk (OH323) -- Quintum A800 codec g729a Call begins without problem but it is closed after nearly 20 seconds. Reason : EndedByRemote When i terminate call at cisco gateways with same configuration there is no problem. But with Quintum i have this problem. Have someone an idea about this problem? Thanks, Ender Erbey conacom GmbH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quintum A800 OH323 problem
Hello, As a start, you can change h245Tunnelling to yes. This will probably solve the problem, as I would receive the messages that you described when I had problems with the H245 negotiation. In addition what are your [codecs] settings in oh323.conf? I assume that you use G729A on the A800 as is shown on the diagram, but what about the settings that you have on the Asterisk side? Also, which version of Asterisk are you using? Best regards, Vlasis. Ender Erbey wrote: Hi, I am using open323 version: 1.13.5 pwlib verison : 1.6.6 OH323 version: 0.6.3b Can this be a configuration problem? Here is my config data: [general] listenAddress=xxx.xxx.xxx.xxx listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=yes h245Tunnelling=no h245inSetup=yes inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=3 libTraceFile=tr.out gatekeeper=DISABLE userInputMode=TONE amaFlags=default accountCode=H323 context=htest Thanks, Ender Erbey conacom GmbH Vlasis Hatzistavrou wrote: Hello, I am passing traffic between Asterisk and A800's with OH323 without problems. No calls are disconnected after 20 seconds. Which version of Asterisk, OH323, pwlib and openh323 are you running? Best regards, Vlasis. Ender Erbey wrote: Hi, I experienced an interesting problem when i try to make such a connection BRI -- Asterisk (OH323) -- Quintum A800 codec g729a Call begins without problem but it is closed after nearly 20 seconds. Reason : EndedByRemote When i terminate call at cisco gateways with same configuration there is no problem. But with Quintum i have this problem. Have someone an idea about this problem? Thanks, Ender Erbey conacom GmbH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP on AVM Fritz with CAPI drivers for SMP RH 9
Hello, Thanks for the suggestion. It seems that AVM supports only the single processor version for this particular card... Though luck I guess... Thanks for the reply anyway. Regards, Vlasis. Thomas Niesel wrote: On Tue, Sep 21, 2004 at 09:00:18PM +0300, Vlasis Chatzistayrou wrote: Hello, I have been wrestling with installing the CAPI drivers for AVM Fritz in order to use chan_capi with Asterisk. I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers (namely: capi4k-utils-2003.06.16-08.mungo.RH9.i686.rpm and kernel-2.4.20-8- avmfcpci-03.11.02-08.mungo.RH9.i686.rpm), but I believe that they support only single processor machines. Check www.avm.de for information about SMP and passive fritz! card-driver4linux Also have a look at ftp.avm.de/cardware/fritzcard.pci/linux/ Check a few (not always the latest) versions. I've already spent too much time with chan_modem which gives me problems (like no audio until the callis answered, or kernel crashes probably because of the isdn4l drivers). So, I can't afford to go back to isdn4linux drivers and the HiSax card that I used. Does anyone have RPMs or source code that I can use for Fritzcard (PCI) and SMP? Thanks in advance for any assistance, Vlasis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with native h323 channel on Asterisk RC2: No early audio and codec negotiation issues
Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2 for example), in which audio would begin before the call was answered. Early audio is useful i order to provide the calling user with remote end ringback as well as recorded announcements, etc. 2) The codec capabilities that Asterisk sends seem strange. No matter which codecs we set in the h323.conf file, G711 is the only codec that is sent in the capabilities. In order to use any other codec, we have to enable only the needed codec and disable all others. Again, this problem did not exist in older * versions, like 0.9.2 and it's limiting the capabilities of Asterisk in H323. Has anyone dealt with this problem successfully? Best regards, -- Vlasis Hatzistavrou. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1
Randy Ackers wrote: Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I agree with Randy, G.723.1 would be extremely useful to many. And since G.723.1 could be used outside of the US from what I understand, it would be very practical if the source code was available for compilation use on Asterisk. Thanks, Vlasis Hatzistavrou. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1
Hello, I have the whole library (G.723.1 and G.723.1b) downloaded from ITU, but it doesn't compile with Asterisk out-of-the-box. So, unless someone else can provide a library which compiles with *, we'll have to tinker with the ITU source code (if it is possible at all). Best regards, Vlasis Hatzistavrou. Stefan de Konink wrote: So simple question, without googling: Where can the g723.1 and g723.1a be found... so a 'EU-patch' can be make. I'm able to host it in Amsterdam. Greetings, Stefan de Konink On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote: Randy Ackers wrote: Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I agree with Randy, G.723.1 would be extremely useful to many. And since G.723.1 could be used outside of the US from what I understand, it would be very practical if the source code was available for compilation use on Asterisk. Thanks, Vlasis Hatzistavrou. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CE certification for Europe
Hello, I'd like to ask if there are any news about CE certification of the E1 boards. I know that the T1 boards are FCC certified but I'd also like to know what is the status for CE certification. Thanks for any input, Vlasis Hatzistavrou. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
