Re: [asterisk-users] 400 calls at g711 how much cpu power
Asterisk max call estimation doesn't scale linearly ... it might in the future with some fixes they're adding. For your application you could use some other open PBX that is known not to have 'Asterisk' limitations. Anyways most people will tell you to simply buy a box and make a test. Noone knows the exact numbers since it's dependant on your kernel version/asterisk version/CPU/motherboard/ethernet card/ memory speed/hdd speed etc. Just make sure the "message" is encoded in G711 ulaw/alaw so there's no transcoding... (use sox) Martin On Wed, Apr 1, 2009 at 10:46 PM, Erick Perez wrote: > We are planning to run an outbound only campaign. A 20-second voice message > will be played to callers and our dialer on machine1 will send to > machine2-asterisk (1.4) instructions to dial 400 calls, play the message and > hang up. This will be done for about 1 million phones. > > The asterisk box will communicate via SIP to a voice carrier. the voice > carrier will then place the calls on pstn. The codec will be g711. So we > will never do any transcoding. > > I have been calculating the CPU power required to do the calls and in > previous posting the usual calculation is about 40MHZ per leg when no > transcoding is involved. > So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. > > Comments? > > -- > > Erick > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 400 calls at g711 how much cpu power
We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP topology hiding
Dear All, Is anyone having luck with using some code for SIP network topology hiding + NAT traversal (SBC functionality) with Asterisk ? I tried OpenSBC but it didn't do NAT from Asterisk to ATA correctly. It's in plans for OpenSIPS but it's not implemented yet ... checked their svn. Martin ___ -- 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] [Zaptel] Why no driver for PCI voice modems?
I wonder why people don't get it ? X100P is a winmodem was and always will be. Martin On Wed, Apr 1, 2009 at 12:26 PM, Tim Nelson wrote: > > If the primary purpose is to drive down cost, why not simply buy any one of > the existing 'Wildcard X100P' clone cards that are everywehere? They're > inexpensive and readily available... > > --Tim ___ -- 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] Extract a MOS value from Asterisk CDR
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine "Irecency", "A" or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
> Sorry for replying for the second time, but this issue is
> interesting for me
> also.
>
> I found such link: http://www.nessoft.com/kb/50
>
> And this: http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
>
>
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> VoIP Billing and Routing Solutions
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
> Leurent
> Sent: 2009 m. balandžio 1 d. 18:15
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
>
> Hello all,
> I'm tring to retrieve a formula to calculate a MOS value from
> Asterisk RTCP
> stats...
> Have you got any idea how to do it?
> Thanks
>
> I'm reading all G.107 ITU docs to retrieve something...
>
> I'm saving the SIP RTCP stats with:
>
> [macro-hangupcall]
> exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
> exten => s,n,ResetCDR(vw)
> exten => s,n,NoCDR()
>
> So I retrieve these values in my MySQL CDR table in order to
> calculate a MOS
>
> value:
> "ssrc
> =
> 592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
> 0;txcount=20734;rlp=0;rtt=0.094000"
> codec used: g711a
>
>
> --
> -- --
> Marc LEURENT
> lf...@leurent.eu
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Sorry for replying for the second time, but this issue is interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this: http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten => s,n,ResetCDR(vw)
exten => s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
"ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000"
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
On Wed, Apr 01, 2009 at 11:16:50PM +0200, Hans Witvliet wrote: > Wasn't that patented under the name of I2CA (Infinite Impropability > Compression Algorithm)... > It was far to technical for me, but afaicr is uses a key with a base of > 42, Or was the exponent 42. can't remember, since then too busy parking > cars ;-( 42.zip ? E.g.: http://www.unforgettable.dk/ -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Addons 1.6.2.0-beta1 Now Available
The Asterisk Development Team is pleased to announce the first beta of Asterisk-Addons 1.6.2.0. Asterisk-Addons 1.6.2.0-beta1 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This beta fixes a several issues with chan_mobile from the chan_mobile refactor branch, and issues related to cdr_addon_mysql and cdr_config_mysql. Additionally, this beta is compatible with Asterisk 1.6.2. For a full list of changes in this beta, please see the ChangeLog: http://svn.digium.com/svn/asterisk-addons/tags/1.6.2.0-beta1/ChangeLog Thank you for your continued support of Asterisk! ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
On Wed, 2009-04-01 at 11:41 -0500, Brent Davidson wrote: > Cary Fitch wrote: > > It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 > > bits each, and that is 2^140^8, a nearly inexhaustible key number which is > > related to audio and video data simultaneously stored on a Google Database, > > which is then sent to the user. > > > > Thus with the 140 byte message, full audio and video can be retrieved. > > > > This is an outgrowth of the data compression program circa about 1992, when > > disks were much smaller than today. A very small compression program would > > infinitely compress data on a disk to allow storage of more data. It was > > only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running > > it once resulted in lots of storage space. It took very little time. Of > > course rewriting the MBR (Master Boot Record) takes very little time. > > > > Recovering the "compressed" data was tough though. > > > > Cary Fitch > > 04/01/09 > > > > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen > > Sent: Wednesday, April 01, 2009 11:09 AM > > To: asterisk-users@lists.digium.com > > Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL > > DRIVERFORASTERISK RELEASED TODAY > > > > On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote: > > > > > I wish we could have this for real > > > > > > > Micro-video-blogging: Limited to 140B ? > > > > > > I thought maybe it used Infinite Monkey Compression where a mathematic > equation whose output over a specified domain would recreate the > data-bits. For those unfamiliar with Infinite Monkey Compression it > was theorized by me a few years ago as an offshoot of Infinite Monkey > Theorem (monkeys, typewriters Shakespeare, etc...). The original > theory was that is an infinite number of monkeys could eventually type > the complete works of Shakespeare through random coincidence then a > random bit generator running for an infinite amount of time would > eventually produce the equivalent bit sequence of any particular piece > of software. Infinity being, well, rather infinite and humans being > mortal and all, infinite runs on a RBG didn't seem like all that great > of an option, so I kept thinking... Then I realized that any file can > be represented by a sequence of numbers. All you have to do is find > the equation that will output those number sequences and you've got a > highly-compressed way to recreate any file. Just send the equation > give it a start and end value and let the computer save the output as > a binary file. Unfortunately I was never able to take IMC beyond the > purely theoretical. > Wasn't that patented under the name of I2CA (Infinite Impropability Compression Algorithm)... It was far to technical for me, but afaicr is uses a key with a base of 42, Or was the exponent 42. can't remember, since then too busy parking cars ;-( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0.7 Now Available
The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a couple of issues with RFC2833 DTMF, and corrects an issue with compiling on CentOS 64-bit platforms. Also, this Asterisk release improves compatibility with T.38 switchovers internally; fixes an issue with Asterisk using poll() on OSX systems when it should not; allows chan_h323 to be built against both OpenH323 and H323Plus libraries (while simplifying the build process); and improve behavior of ast_answer() which was problematic for T.38 re-INVITES and other sorts of channel operations. Additionally, other bugs have been resolved in this release. For a summary of the changes in this release, please see the release summary: http://svn.digium.com/svn/asterisk/tags/1.6.0.7/asterisk-1.6.0.7-summary.txt For a full list of changes in this release, please see the ChangeLog: http://svn.digium.com/svn/asterisk/tags/1.6.0.7/ChangeLog The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help! * Allow H.323 Plus library to be used in addition to the OpenH323 library - Closes issue #11261. Submitted by vhatz. Patched by jthurman. * Make the sip_standard_port function more granular by allowing separate type and port arguments. - Closes issue #12761. Reported and patched by asbestoshead. * Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. - Closes issue #14538. Reported and patched by KNK. * Remove duplicate 'k' and 'K' Dial options. - Closes issue #14601. Reported and patched by alecdavis. Thank you for your continued support of Asterisk! ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
On Wed, Apr 1, 2009 at 3:18 AM, Olle E. Johansson wrote: What a shame about the loss of chan_hype. I was really hoping to build a .com around it. At least I'm feeling better since starting the placebo treatment for my allergies. ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
Computing used to be fun. Now we have to make the buttons on the phone blink, even if the manufacturer didn't put an LED or circuit behind the button. :-) CF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of zoach...@securax.org Sent: Wednesday, April 01, 2009 3:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY For those looking for the faq on that page :) http://unix.derkeiler.com/Newsgroups/comp.os.vms/2003-07/2406.html Tzafrir Cohen wrote: > On Wed, Apr 01, 2009 at 11:27:17AM -0500, Cary Fitch wrote: > >> It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 >> bits each, and that is 2^140^8, a nearly inexhaustible key number which is >> related to audio and video data simultaneously stored on a Google Database, >> which is then sent to the user. >> >> Thus with the 140 byte message, full audio and video can be retrieved. >> >> This is an outgrowth of the data compression program circa about 1992, when >> disks were much smaller than today. A very small compression program would >> infinitely compress data on a disk to allow storage of more data. It was >> only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running >> it once resulted in lots of storage space. It took very little time. Of >> course rewriting the MBR (Master Boot Record) takes very little time. >> >> Recovering the "compressed" data was tough though. >> > > There were some later implementations of that idea. Here's a rather > efficient one: > > http://web.archive.org/web/20010405094403/http://lzip.sourceforge.net/ > > ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
For those looking for the faq on that page :) http://unix.derkeiler.com/Newsgroups/comp.os.vms/2003-07/2406.html Tzafrir Cohen wrote: > On Wed, Apr 01, 2009 at 11:27:17AM -0500, Cary Fitch wrote: > >> It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 >> bits each, and that is 2^140^8, a nearly inexhaustible key number which is >> related to audio and video data simultaneously stored on a Google Database, >> which is then sent to the user. >> >> Thus with the 140 byte message, full audio and video can be retrieved. >> >> This is an outgrowth of the data compression program circa about 1992, when >> disks were much smaller than today. A very small compression program would >> infinitely compress data on a disk to allow storage of more data. It was >> only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running >> it once resulted in lots of storage space. It took very little time. Of >> course rewriting the MBR (Master Boot Record) takes very little time. >> >> Recovering the "compressed" data was tough though. >> > > There were some later implementations of that idea. Here's a rather > efficient one: > > http://web.archive.org/web/20010405094403/http://lzip.sourceforge.net/ > > ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNELDRIVERFORASTERISK RELEASED TODAY
If you had done it once more you would have had it down to half a bit.
Quantum computing?
j
On Wed, 1 Apr 2009, Cary Fitch wrote:
> Yeah got it down to 1 bit that way.
>
>
>
> exten byte1 => (dataflag=(${byte1}:bit1)?had-data:didn't-have-data))
>
>
>
> If dataflag returns "had-data" recovering the data you call and parse an
> external subroutine the same size and composition of the original data.
>
> Otherwise no external routine is needed.
>
> Cary
>
>
>
>
>
>
>
> _
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
> Victor
> Sent: Wednesday, April 01, 2009 3:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW
> CHANNELDRIVERFORASTERISK RELEASED TODAY
>
>
>
> Duuh guys - it's so easy. Ever thought of simply compressing the compressed
> data AGAIN???
>
> Do that the necessary amount of times and - tadaa - it's done.
>
> Chris
>
> 2009/4/1 Brent Davidson
>
> Cary Fitch wrote:
>
> It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
> bits each, and that is 2^140^8, a nearly inexhaustible key number which is
> related to audio and video data simultaneously stored on a Google Database,
> which is then sent to the user.
>
> Thus with the 140 byte message, full audio and video can be retrieved.
>
> This is an outgrowth of the data compression program circa about 1992, when
> disks were much smaller than today. A very small compression program would
> infinitely compress data on a disk to allow storage of more data. It was
> only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running
> it once resulted in lots of storage space. It took very little time. Of
> course rewriting the MBR (Master Boot Record) takes very little time.
>
> Recovering the "compressed" data was tough though.
>
> Cary Fitch
> 04/01/09
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
> Sent: Wednesday, April 01, 2009 11:09 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL
> DRIVERFORASTERISK RELEASED TODAY
>
> On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote:
>
>
> I wish we could have this for real
>
>
> Micro-video-blogging: Limited to 140B ?
>
>
>
>
>
> I thought maybe it used Infinite Monkey Compression where a mathematic
> equation whose output over a specified domain would recreate the data-bits.
> For those unfamiliar with Infinite Monkey Compression it was theorized by me
> a few years ago as an offshoot of Infinite Monkey Theorem (monkeys,
> typewriters Shakespeare, etc...). The original theory was that is an
> infinite number of monkeys could eventually type the complete works of
> Shakespeare through random coincidence then a random bit generator running
> for an infinite amount of time would eventually produce the equivalent bit
> sequence of any particular piece of software. Infinity being, well, rather
> infinite and humans being mortal and all, infinite runs on a RBG didn't seem
> like all that great of an option, so I kept thinking... Then I realized
> that any file can be represented by a sequence of numbers. All you have to
> do is find the equation that will output those number sequences and you've
> got a highly-compressed way to recreate any file. Just send the equation
> give it a start and end value and let the computer save the output as a
> binary file. Unfortunately I was never able to take IMC beyond the purely
> theoretical.
>
>
> ___
> -- 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
>
>
>
>
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Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNELDRIVERFORASTERISK RELEASED TODAY
Yeah got it down to 1 bit that way.
exten byte1 => (dataflag=(${byte1}:bit1)?had-data:didn't-have-data))
If dataflag returns "had-data" recovering the data you call and parse an
external subroutine the same size and composition of the original data.
Otherwise no external routine is needed.
Cary
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Victor
Sent: Wednesday, April 01, 2009 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW
CHANNELDRIVERFORASTERISK RELEASED TODAY
Duuh guys - it's so easy. Ever thought of simply compressing the compressed
data AGAIN???
Do that the necessary amount of times and - tadaa - it's done.
Chris
2009/4/1 Brent Davidson
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that is 2^140^8, a nearly inexhaustible key number which is
related to audio and video data simultaneously stored on a Google Database,
which is then sent to the user.
Thus with the 140 byte message, full audio and video can be retrieved.
This is an outgrowth of the data compression program circa about 1992, when
disks were much smaller than today. A very small compression program would
infinitely compress data on a disk to allow storage of more data. It was
only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running
it once resulted in lots of storage space. It took very little time. Of
course rewriting the MBR (Master Boot Record) takes very little time.
Recovering the "compressed" data was tough though.
Cary Fitch
04/01/09
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Wednesday, April 01, 2009 11:09 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL
DRIVERFORASTERISK RELEASED TODAY
On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote:
I wish we could have this for real
Micro-video-blogging: Limited to 140B ?
I thought maybe it used Infinite Monkey Compression where a mathematic
equation whose output over a specified domain would recreate the data-bits.
For those unfamiliar with Infinite Monkey Compression it was theorized by me
a few years ago as an offshoot of Infinite Monkey Theorem (monkeys,
typewriters Shakespeare, etc...). The original theory was that is an
infinite number of monkeys could eventually type the complete works of
Shakespeare through random coincidence then a random bit generator running
for an infinite amount of time would eventually produce the equivalent bit
sequence of any particular piece of software. Infinity being, well, rather
infinite and humans being mortal and all, infinite runs on a RBG didn't seem
like all that great of an option, so I kept thinking... Then I realized
that any file can be represented by a sequence of numbers. All you have to
do is find the equation that will output those number sequences and you've
got a highly-compressed way to recreate any file. Just send the equation
give it a start and end value and let the computer save the output as a
binary file. Unfortunately I was never able to take IMC beyond the purely
theoretical.
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Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
Duuh guys - it's so easy. Ever thought of simply compressing the compressed data AGAIN??? Do that the necessary amount of times and - tadaa - it's done. Chris 2009/4/1 Brent Davidson > Cary Fitch wrote: > > It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 > bits each, and that is 2^140^8, a nearly inexhaustible key number which is > related to audio and video data simultaneously stored on a Google Database, > which is then sent to the user. > > Thus with the 140 byte message, full audio and video can be retrieved. > > This is an outgrowth of the data compression program circa about 1992, when > disks were much smaller than today. A very small compression program would > infinitely compress data on a disk to allow storage of more data. It was > only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running > it once resulted in lots of storage space. It took very little time. Of > course rewriting the MBR (Master Boot Record) takes very little time. > > Recovering the "compressed" data was tough though. > > Cary Fitch > 04/01/09 > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com > ] On Behalf Of Tzafrir Cohen > Sent: Wednesday, April 01, 2009 11:09 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL > DRIVERFORASTERISK RELEASED TODAY > > On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote: > > > I wish we could have this for real > > > Micro-video-blogging: Limited to 140B ? > > > > > I thought maybe it used Infinite Monkey Compression where a mathematic > equation whose output over a specified domain would recreate the data-bits. > For those unfamiliar with Infinite Monkey Compression it was theorized by me > a few years ago as an offshoot of Infinite Monkey Theorem (monkeys, > typewriters Shakespeare, etc...). The original theory was that is an > infinite number of monkeys could eventually type the complete works of > Shakespeare through random coincidence then a random bit generator running > for an infinite amount of time would eventually produce the equivalent bit > sequence of any particular piece of software. Infinity being, well, rather > infinite and humans being mortal and all, infinite runs on a RBG didn't seem > like all that great of an option, so I kept thinking... Then I realized > that any file can be represented by a sequence of numbers. All you have to > do is find the equation that will output those number sequences and you've > got a highly-compressed way to recreate any file. Just send the equation > give it a start and end value and let the computer save the output as a > binary file. Unfortunately I was never able to take IMC beyond the purely > theoretical. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Context Confusion
Okay, I am not understanding if I have this correct or not. I have a requirement to allow guests into a PBX from different domains. However, I can not allow the guests into the default context because each domain has its own IVR. So I end up setting the domain context. I also need to provide separate contexts for different sip users (different dial groups). Small system, few users, so it doesn't make sense to create separate Asterisk boxes (cost wise and support) and some of the prompts are similar. Same company, different micro departments and web domains. Should need to either. If I set the user context to "user1" and have set a domain context set to "guests1" in sip.conf, the system is ignoring the "user1" context. An incoming call (from the code) will be force the context to "guests1" and not have the "user1". I quote: /* If we have a context defined, overwrite the original context */ For example, in sip.conf: [general] context=fromsip domain=domain1.tld,guests1 domain=domain2.tld,guests2 [userA] context=user1 It would seem to me, that if the context was NOT set in the SIP entry, and a domain context was available, only then would you replace the context. To me, I would go from micro to macro definition and not jump around. So we would have peer, domain, general in the SIP context hierarchy. Instead we have domain, peer, general. What am I missing about why this is setup this way (other than that is the way it has always been)? Looking for some instruction here to wrap my head around this better. As stands now, I believe I have to set all the phones up to a domain without a context to allow the local context to be used. Is that correct? ___ -- 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] iax2 not registering at startup, works on reload
On Tue, Mar 31, 2009 at 10:27:45AM +0100, Steve Davies wrote: > > Most commonly, if DNS is not ready to resolve a hostname, IAX can > stall and/or fail to register. DNS was the cause. Replacing the hostname with its IP address fixed it. Thanks! -Yahya ___ -- 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] Trunk SIP and configuration
Hello,
I don't speak english very well but i think.
[operador]
qualify=yes
nat=yes
host=192.168.700.50
insecure=invite,port
canreinvite=no
context=default
disallow=all
allow=ulaw
allow=g729
in your extensions.conf
exten => _00X,1, Dial (SIP/operador/${EXTEN},60,tT)
Best Regards
Carlos Rojas
On Wed, Apr 1, 2009 at 10:45 AM, ludo perrot wrote:
> hello,
>
> I am beginning to asterisk.
> I have a sip trunk access to operator and VPN access with operator.
> i booked 10 sda numbers.
>
> IP adress asterisk : 192.168.600.1
> IP adress operator : 192.168.700.50
> i can ping on 192.168.700.50
>
>
> # cat sip.conf
> [general]
> context=default
> srvlookup=yes
> port = 5060
> disallow=all
> allow=gsm
> allow=alaw
> allow=ulaw
>
> [1000]
> username=1000
> type=friend
> qualify=yes
> secret=3615
> nat=no
> host=192.168.600.3
> canreinvite=no
> context=appels_entrants
>
> [Catherine]
> usename=1010
> type=friend
> qualify=yes
> secret=5768
> nat=yes
> host=192.168.600.4
> canreinvite=yes
> context=default
> disallow=all
> allow=ulaw
>
> # extensions.conf
> exten => _00X,1, Dial (SIP/192.168.700.50/${EXTEN})
>
> How do I configure IP operator ?
> I have 10 numbers sda. Where do I configure sda numbers ?
>
> Thanks.
> Ludovic
>
>
>
>
> ___
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>
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Re: [asterisk-users] Remote host can't match request CANCEL to call
1 apr 2009 kl. 16.24 skrev Grygoriy Dobrovolskyy: > > > 2009/4/1 Shaun Wingrin > Hi, > > Why does this warning occur and what are the implications of it? I'm > concerned about calls never getting hung up.! > > chan_sip.c:12890 handle_response: Remote host can't match request > CANCEL to call '2f197e56611061a678c13b881b269...@411.2.139.106'. > Giving up. > > Tx > > > Hello > It's other end who is not aware if the call leg for that cancel, it > is happening when some provider missconfigured the load balancing > stuff for example, or call leg allready was destroyed for any reason. > This was actually caused by a bug in some version of Asterisk 1.4, which I fixed. If you update it should not happen. /O --- * Olle E. Johansson - o...@edvina.net * Asterisk Training http://edvina.net/training/ ___ -- 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] [Zaptel] Why no driver for PCI voice modems?
Fred wrote: > Hello > > Considering how cheap PCI modems are compared to even the cheapest > PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering > why Zaptel can't be used with those to connect an Asterisk server to > a POTS line for low-level use? It just seems overkill for SOHO users > who only get a few calls a day. Hi Fred, just purchase an X100[p] clone on ebay. I bought one last year from a seller in the USA and it cost me about £17 (GBP) including shipping. Using Zaptel and OSLEC it is absolutely fine. HTH Alan PS - I did a quick search and here is the type of thing I mean. I have no idea of the seller or this particular board but you get the idea I'm sure: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=220380070996 ___ -- 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] [Zaptel] Why no driver for PCI voice modems?
Tim Nelson wrote: > - "Wilton Helm" wrote: > > > >If half-duplex audio is good enough for you, sure. > > You've lost me there. I am not aware of a modem that is for sale > today that is half duplex. (OK some support a couple of minor half > duplex modes). All state of the art modem protocols send and receive > simultaneously using the full 300 - 3000 Hz bandwidth in both > directions with adaptive equalization and echo cancellation to make it > work, which is pretty much what a voice circuit need. There are two > differences: 1) The response and quality of a current modem must be > considerably higher than what is needed for voice use or it would > never achieve the throughput expected of it, and 2) the adaptive > equalization algorithm is designed around modem specific techniques. > The latter is (especially for a softmodem) a software issue, not a > hardware limitation. The hardware modems are usually half-duplex. If you use the hardware of winmodems directly, bypassing their normal drivers, you can get a good quality bidirectional channel. > > >Only a fraction of the hardware available is actually capable of full > duplex audio. > > > > Absolutely not the case. Particularly the softmodems (the most > inexpensive) contain little else than what is required for placing and > answering full duplex audio calls. Everything else is in the driver. > The OP is 100% correct, that they would be an excellent candidate for > FXO use in low volume applications. Its only really the winmodems that are of interest, so you are quite right. > > > >What it really comes down to is a value proposition: > > Quite true. This is the real issue. As mentioned, these drivers > require considerable skill and knowledge to write. While there is no > doubt that the result would be very cost effective, the business model > is lacking. The modem manufacturer is going to see the potential > market for this as somewhere down in the noise compared to their > normal modem sales, so isn't inclined to invest. A third party > developer with the skills would have a difficult time recouping > development costs (let alone any profit) because they don't control > the hardware, and therefore have no leverage. A user with enough > volume to justify paying for the development (or doing it if they had > the skill) probably has enough volume to use T1s instead. If everyone > that could benefit from using a modem card were to pitch in $10 > towards the development, it would probably be quite possible. But how > to make that happen? Its straightforward to achieve, but nobody bothers. The Linux drivers for most winmodems have the DSP (which you don't need) as a binary blob, and the kernel driver (which you'll need to modify) as source code. As a modem they normally run at 9600 samples/second. Most of the chips can be programmed for 8000 samples/second, though, so they'll do what you need. Use them with OSLEC, and you could get great results. There are only a few suppliers of these winmodem chips - some USB and some PCI. You wouldn't need a lot of drivers to cover practically the whole market. As I said. Its all possible. The necessary hardware info is mostly out there in downloadable source code. Its just that nobody has bothered. Regards, Steve ___ -- 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] [Zaptel] Why no driver for PCI voice modems?
I don't think a off the shelf modem has the necessary DSPs to convert voice to codecthat is why a Voice Gateway/Analog Telephony Adapter or FXO/FXS cards exist instead of modem having a second life. I do recall a few that worked as a answering machine allowing your home computer to answer calls ,etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wilton Helm Sent: Wednesday, April 01, 2009 1:06 PM To: Asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems? >If half-duplex audio is good enough for you, sure. You've lost me there. I am not aware of a modem that is for sale today that is half duplex. (OK some support a couple of minor half duplex modes). All state of the art modem protocols send and receive simultaneously using the full 300 - 3000 Hz bandwidth in both directions with adaptive equalization and echo cancellation to make it work, which is pretty much what a voice circuit need. There are two differences: 1) The response and quality of a current modem must be considerably higher than what is needed for voice use or it would never achieve the throughput expected of it, and 2) the adaptive equalization algorithm is designed around modem specific techniques. The latter is (especially for a softmodem) a software issue, not a hardware limitation. >Only a fraction of the hardware available is actually capable of full duplex audio. Absolutely not the case. Particularly the softmodems (the most inexpensive) contain little else than what is required for placing and answering full duplex audio calls. Everything else is in the driver. The OP is 100% correct, that they would be an excellent candidate for FXO use in low volume applications. >What it really comes down to is a value proposition: Quite true. This is the real issue. As mentioned, these drivers require considerable skill and knowledge to write. While there is no doubt that the result would be very cost effective, the business model is lacking. The modem manufacturer is going to see the potential market for this as somewhere down in the noise compared to their normal modem sales, so isn't inclined to invest. A third party developer with the skills would have a difficult time recouping development costs (let alone any profit) because they don't control the hardware, and therefore have no leverage. A user with enough volume to justify paying for the development (or doing it if they had the skill) probably has enough volume to use T1s instead. If everyone that could benefit from using a modem card were to pitch in $10 towards the development, it would probably be quite possible. But how to make that happen? Wilton - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] [Zaptel] Why no driver for PCI voice modems?
- "Wilton Helm" wrote: > >If half-duplex audio is good enough for you, sure. You've lost me there. I am not aware of a modem that is for sale today that is half duplex. (OK some support a couple of minor half duplex modes). All state of the art modem protocols send and receive simultaneously using the full 300 - 3000 Hz bandwidth in both directions with adaptive equalization and echo cancellation to make it work, which is pretty much what a voice circuit need. There are two differences: 1) The response and quality of a current modem must be considerably higher than what is needed for voice use or it would never achieve the throughput expected of it, and 2) the adaptive equalization algorithm is designed around modem specific techniques. The latter is (especially for a softmodem) a software issue, not a hardware limitation. >Only a fraction of the hardware available is actually capable of full duplex >audio. > Absolutely not the case. Particularly the softmodems (the most inexpensive) contain little else than what is required for placing and answering full duplex audio calls. Everything else is in the driver. The OP is 100% correct, that they would be an excellent candidate for FXO use in low volume applications. >What it really comes down to is a value proposition: Quite true. This is the real issue. As mentioned, these drivers require considerable skill and knowledge to write. While there is no doubt that the result would be very cost effective, the business model is lacking. The modem manufacturer is going to see the potential market for this as somewhere down in the noise compared to their normal modem sales, so isn't inclined to invest. A third party developer with the skills would have a difficult time recouping development costs (let alone any profit) because they don't control the hardware, and therefore have no leverage. A user with enough volume to justify paying for the development (or doing it if they had the skill) probably has enough volume to use T1s instead. If everyone that could benefit from using a modem card were to pitch in $10 towards the development, it would probably be quite possible. But how to make that happen? Wilton If the primary purpose is to drive down cost, why not simply buy any one of the existing 'Wildcard X100P' clone cards that are everywehere? They're inexpensive and readily available... --Tim ___ -- 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] Asterisk doesn't relay remote MOH during hold
Hi Tony I can see where you guys are coming from on this and have already enumerated your argument in my own email. But there are very real reasons for a PBX to signal the hold even when it wants to send its own MOH: 1. Bandwidth: under your scheme the PBX would continue to receive bandwidth-consuming media without using it. 2. Privacy: the far-end has an expectation of privacy while on hold and should have the option to mute automatically when held. 3. Feature richness: signalling the hold enables such innovative features such as reverse hold. 4. ISDN interworking: ISDN supports this and SIP should be compatible with that (as per standard ITU-T Q.1912.5) Also, can you explain why the PBX would use a=sendonly but not dispatch media. Why not a=inactive for that case? > IMHO, PBX-A would be broken if it passed this along the Hold message to > downstream and then started servicing the MOH itself Remember it is not a hold message, it is a media attribute and we are discussing how that should be interpreted within the context of the hold feature in traditional telephony. I would also like to point out in my defence that there are several telephone systems in the field which behave as I described (Nortel BCM50, Aastra Intelligate, Mitel 3300 to name a few). Regards, Richard > I have to agree with Kevin on this one. > > I fail to understand how you have a PBX-A talking to Asterisk talking to > PBX-B and the PBX-A placing the call on hold. Typically you should have a > Client/Phone to PBX-A to Asterisk to PBX-B to Client/Phone/VoiceMail. > > If the Client signals Hold, the PBX should NOT be passing that Hold status on > but transition audio stream from Client to MOH (assuming MOH is handled). > Asterisk shouldn't notice a thing except more RTP packets (or less if it is > my teenage daughter on the phone as the case may be). > > IMHO, PBX-A would be broken if it passed this along the Hold message to > downstream and then started servicing the MOH itself on the RTP stream. That > just doesn't make sense. > > Now if PBX-A were not a PBX and were a SIP Router, and the SIP Router was > attempting this, I can see how it would Re-Invite, but it shouldn't pass the > hold status onto Asterisk. > > Need some clarity here. > > Tony Plack ___ -- 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] [Zaptel] Why no driver for PCI voice modems?
>If half-duplex audio is good enough for you, sure. You've lost me there. I am not aware of a modem that is for sale today that is half duplex. (OK some support a couple of minor half duplex modes). All state of the art modem protocols send and receive simultaneously using the full 300 - 3000 Hz bandwidth in both directions with adaptive equalization and echo cancellation to make it work, which is pretty much what a voice circuit need. There are two differences: 1) The response and quality of a current modem must be considerably higher than what is needed for voice use or it would never achieve the throughput expected of it, and 2) the adaptive equalization algorithm is designed around modem specific techniques. The latter is (especially for a softmodem) a software issue, not a hardware limitation. >Only a fraction of the hardware available is actually capable of full duplex >audio. Absolutely not the case. Particularly the softmodems (the most inexpensive) contain little else than what is required for placing and answering full duplex audio calls. Everything else is in the driver. The OP is 100% correct, that they would be an excellent candidate for FXO use in low volume applications. >What it really comes down to is a value proposition: Quite true. This is the real issue. As mentioned, these drivers require considerable skill and knowledge to write. While there is no doubt that the result would be very cost effective, the business model is lacking. The modem manufacturer is going to see the potential market for this as somewhere down in the noise compared to their normal modem sales, so isn't inclined to invest. A third party developer with the skills would have a difficult time recouping development costs (let alone any profit) because they don't control the hardware, and therefore have no leverage. A user with enough volume to justify paying for the development (or doing it if they had the skill) probably has enough volume to use T1s instead. If everyone that could benefit from using a modem card were to pitch in $10 towards the development, it would probably be quite possible. But how to make that happen? Wilton ___ -- 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] Avoid compression with g.729/gsm/etc.
> Regarding compression with g.729/gsm/etc. and Asterisk > > If we convert all the voice files to the corresponding format g.729/gsm/etc. > and we send digits using RFC 3261 and we do not need silence detection, is > there still a need to decompress the media stream ? > > If doable how to make sure this will work without compression/decompression ? > > I believe that Asterisk by default unpackages/repackages the stream. If you are looking for RTP pass-through, you are needing a RTP Proxy or SIP Reinvite and not Asterisk. Look at kamailio.org and RTP Proxy with Asterisk as the VoiceMail/Media Server. ___ -- 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] Asterisk doesn't relay remote MOH during hold
> Ok, this is where it gets interesting. Consider the case of a PBX > which has its own MOH source and is talking via Asterisk to another > PBX. > > If that PBX wants to put the call on hold while sending its own MOH, > you would probably argue that it should not send a re-INIVTE at all, > but should simply replace the outbound audio stream with its MOH and > discard the inbound audio stream. I have to agree with Kevin on this one. I fail to understand how you have a PBX-A talking to Asterisk talking to PBX-B and the PBX-A placing the call on hold. Typically you should have a Client/Phone to PBX-A to Asterisk to PBX-B to Client/Phone/VoiceMail. If the Client signals Hold, the PBX should NOT be passing that Hold status on but transition audio stream from Client to MOH (assuming MOH is handled). Asterisk shouldn't notice a thing except more RTP packets (or less if it is my teenage daughter on the phone as the case may be). IMHO, PBX-A would be broken if it passed this along the Hold message to downstream and then started servicing the MOH itself on the RTP stream. That just doesn't make sense. Now if PBX-A were not a PBX and were a SIP Router, and the SIP Router was attempting this, I can see how it would Re-Invite, but it shouldn't pass the hold status onto Asterisk. Need some clarity here. Tony Plack ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
On Wed, Apr 01, 2009 at 11:27:17AM -0500, Cary Fitch wrote: > It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 > bits each, and that is 2^140^8, a nearly inexhaustible key number which is > related to audio and video data simultaneously stored on a Google Database, > which is then sent to the user. > > Thus with the 140 byte message, full audio and video can be retrieved. > > This is an outgrowth of the data compression program circa about 1992, when > disks were much smaller than today. A very small compression program would > infinitely compress data on a disk to allow storage of more data. It was > only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running > it once resulted in lots of storage space. It took very little time. Of > course rewriting the MBR (Master Boot Record) takes very little time. > > Recovering the "compressed" data was tough though. There were some later implementations of that idea. Here's a rather efficient one: http://web.archive.org/web/20010405094403/http://lzip.sourceforge.net/ -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
Cary Fitch wrote: It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 bits each, and that is 2^140^8, a nearly inexhaustible key number which is related to audio and video data simultaneously stored on a Google Database, which is then sent to the user. Thus with the 140 byte message, full audio and video can be retrieved. This is an outgrowth of the data compression program circa about 1992, when disks were much smaller than today. A very small compression program would infinitely compress data on a disk to allow storage of more data. It was only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running it once resulted in lots of storage space. It took very little time. Of course rewriting the MBR (Master Boot Record) takes very little time. Recovering the "compressed" data was tough though. Cary Fitch 04/01/09 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Wednesday, April 01, 2009 11:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote: I wish we could have this for real Micro-video-blogging: Limited to 140B ? I thought maybe it used Infinite Monkey Compression where a mathematic equation whose output over a specified domain would recreate the data-bits. For those unfamiliar with Infinite Monkey Compression it was theorized by me a few years ago as an offshoot of Infinite Monkey Theorem (monkeys, typewriters Shakespeare, etc...). The original theory was that is an infinite number of monkeys could eventually type the complete works of Shakespeare through random coincidence then a random bit generator running for an infinite amount of time would eventually produce the equivalent bit sequence of any particular piece of software. Infinity being, well, rather infinite and humans being mortal and all, infinite runs on a RBG didn't seem like all that great of an option, so I kept thinking... Then I realized that any file can be represented by a sequence of numbers. All you have to do is find the equation that will output those number sequences and you've got a highly-compressed way to recreate any file. Just send the equation give it a start and end value and let the computer save the output as a binary file. Unfortunately I was never able to take IMC beyond the purely theoretical. ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 bits each, and that is 2^140^8, a nearly inexhaustible key number which is related to audio and video data simultaneously stored on a Google Database, which is then sent to the user. Thus with the 140 byte message, full audio and video can be retrieved. This is an outgrowth of the data compression program circa about 1992, when disks were much smaller than today. A very small compression program would infinitely compress data on a disk to allow storage of more data. It was only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running it once resulted in lots of storage space. It took very little time. Of course rewriting the MBR (Master Boot Record) takes very little time. Recovering the "compressed" data was tough though. Cary Fitch 04/01/09 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Wednesday, April 01, 2009 11:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote: > I wish we could have this for real Micro-video-blogging: Limited to 140B ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FORASTERISK RELEASED TODAY
On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote: > I wish we could have this for real Micro-video-blogging: Limited to 140B ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FORASTERISK RELEASED TODAY
I wish we could have this for real - Original Message - From: "Olle E. Johansson" To: "Asterisk Non-Commercial Discussion Users Mailing List -" Sent: Wednesday, April 01, 2009 10:18 AM Subject: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FORASTERISK RELEASED TODAY >* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND > VIDEO TO MICROBLOGGING! > > In a surprising move, Digium in partnership with Edvina today released > a new channel driver for Asterisk, chan_tweet. The driver connects > seamlessly to several microblogging platforms, including Twitter, > Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of > this new module is to add audio and video capabilities to > microblogging, making the popular microblogging networks a new > platform for VoIP and IP realtime communication. > > - "I have seen that the microblogging solutions building on the social > network infrastructure have had enourmous unexploited capabilities", > says Mill Biller at Digium, "I've used it for a long time both > personally and for the company and we realized early that by adding > IAX2 support, we could now take these platforms one giant leap forward > by adding realtime multimedia. I can now spend evenings chit-chatting > in audio and HD-resolution video with all my audience around the world > instead of sending short text messages. It's truly awsome!" > > Digium contracted Edvina in Sweden, a well-known company in the > Asterisk community and long-term Digium business partner, to build > this solution. Edvina has many years of experience in building large- > scale IAX2 networks, as well as doing development on the IAX2 support > in Asterisk. > > - "IAX2 recently was published in an IETF RFC and we're pushing it > heavily in all VoIP forums." says Olle Johansson of Edvina, "We're > hoping that the IAX2FORUM will get a lot of new members that are > willing to adopt this technology for their intranets, microblogging > services and VoIP infrastructures. In the coming month, we will > present more information about new partners with more than 100K users > that are going to switch from old technologies, like Hype, SIP and H. > 323. All of these protocols failed, either because they where > proprietary or simply became too complex. SIP currently has more than > 5.000 pages of documents describing all the features of the protocol > and there's no single implementation of all of these to test with. > Considering the protocol being over 10 years old, this is a sad story." > > - "We've done our best to fix the Asterisk SIP channel support for > customers, but the customer base has been shrinking as more and more > converted their networks to IAX2 and now, there's simply no one > interested in us doing that work. We've stated over and over again > that the SIP channel in Asterisk is broken and no one can prove us > right or wrong, because the protocol is just too complex." > > * The Microblogmedia platform > -- > The Microblogmedia(TM) platform, developed by Digium and Edvina, let's > users use any microblogging network to set up multimedia sessions. By > compressing an IAX2 call setup event in the microblog message, web > browsers and clients will connect automatically peer-2-peer if > possible, or through the MicroBlogMediaRelay network that supports > seamless NAT and firewall traversal by using automatic IPv6 tunnels. > > Asterisk 1.6.3, released later this month, will support this feature > in the IAX2, H.323 and maybe in the old SIP channel (that is now > marked deprecated). There is work on adding this feature to ISDN > calls, by using messages in the D-channel for tunneling the IAX2 call > setup messages. Digium's VoxSwitch will support this feature in the > next release, planned for q3 2009. > > * Ending the Hype project > --- > In the same press release, Sock Stevens, product manager at Digium > finally acknowledged that the Hype channel driver that was launched at > Astricon 2008 will not be released after all. > - "We found only one partner to test interoperability with, and that's > not enough to make sure the channel driver being compatible with the > protocol. And the protocol wasn't published in any RFC at all, or any > other document. So we finally gave up. We're now dedicating resources > for the new chan_tweet project and enhancing presence support in our > IAX2 solution. With the installed base of IAX2 and the new > MicroBlogMedia platform, this will be an even more impressive > solution, reaching millions of IAX2 users in the enterprise as well as > public sector and homes." > > * Technichal factoids > > - chan_tweet is the result of the project labelled "Codename > orangepeel" amongst the development team and builds on the new > "Pinemango" architecture. This is the first channel driver not > connecting directly to the Asterisk core, but to the Pinemango AP
[asterisk-users] Trunk SIP and configuration
hello,
I am beginning to asterisk.
I have a sip trunk access to operator and VPN access with operator.
i booked 10 sda numbers.
IP adress asterisk : 192.168.600.1
IP adress operator : 192.168.700.50
i can ping on 192.168.700.50
# cat sip.conf
[general]
context=default
srvlookup=yes
port = 5060
disallow=all
allow=gsm
allow=alaw
allow=ulaw
[1000]
username=1000
type=friend
qualify=yes
secret=3615
nat=no
host=192.168.600.3
canreinvite=no
context=appels_entrants
[Catherine]
usename=1010
type=friend
qualify=yes
secret=5768
nat=yes
host=192.168.600.4
canreinvite=yes
context=default
disallow=all
allow=ulaw
# extensions.conf
exten => _00X,1, Dial (SIP/192.168.700.50/${EXTEN})
How do I configure IP operator ?
I have 10 numbers sda. Where do I configure sda numbers ?
Thanks.
Ludovic
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Re: [asterisk-users] login-logout asterisk
Here is a simple control for what you are asking:
Exten => s,1,noop(Dial Long Distance #)
exten => s,n,Set(LDACCESS=${DB(LD/Access)})
exten => s,n(readacct),Read(digitacc,record/entercode,8,skip,1,10])
exten => s,n,Gotoif($["${LEN(${digitacc})}" < "4"]?readacct)
exten => s,n,Gotoif($["${digitacc}" < "${LDACCESS}"]?readacct)
exten => s,n,Dial(Tech/1,ww1,60)
This reads the access code in from the Asterisk DB, plays a message for user
to enter it and dials once a matching entry is made.
You have to create the value in the Asterisk DB and record the message.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Wednesday, April 01, 2009 10:26 AM
To: asterisk
Subject: [asterisk-users] login-logout asterisk
Hello,
In our previous PBX we have an option to turn off or on outside calls with
a pincode..
Like, user is able to get calls or dial local lines by default, but when
he/she uses a password entrance via dtmf, he can dial long distance calls
etc.And at anytime he can logoff from outside call permit..
So is it possible to do smthing like this on asterisk..
A limited profile which needs sip password of the user again to dial long
distance calls for example.
Thanks..
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Check this:
http://www.voip-info.org/wiki/index.php?page=Call+Quality+Metrics
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten => s,n,ResetCDR(vw)
exten => s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
"ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000"
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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[asterisk-users] login-logout asterisk
Hello, In our previous PBX we have an option to turn off or on outside calls with a pincode.. Like, user is able to get calls or dial local lines by default, but when he/she uses a password entrance via dtmf, he can dial long distance calls etc.And at anytime he can logoff from outside call permit.. So is it possible to do smthing like this on asterisk.. A limited profile which needs sip password of the user again to dial long distance calls for example. Thanks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten => s,n,ResetCDR(vw)
exten => s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
"ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=20734;rlp=0;rtt=0.094000"
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Re: [asterisk-users] Forking
hh174 wrote: > Hello all, > > Probably a bad news for all... > > The Undercompetent Olle E Johansson decided to leave the asterisk team > to create his own Voip server. > Oh Ma GOSH! I guess I'll trash all my installs and move over to Avaya! *snicker* -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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] Forking
At least fake your from email to make it believable.. hh174 wrote: > Hello all, > > Probably a bad news for all... > > The Undercompetent Olle E Johansson decided to leave the asterisk team > to create his own Voip server. > The server will be called Minisk (due probably to his poor competence in > Voip). > Following that, Digium decides to stop any development on Asterisk and > joined the Skype team to recreate a brand new paying licensed software. > Any development will be halted for asterisk > > It seems that Mr Johansson has was poisoned by a fish in Brussels during > the last Fosdem. > > We all hope qur Mr Johansson will quickly heal and return to the team of > asterisk. > > Kind regards, > > Olivier Taylor > > > > ___ > -- 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] Remote host can't match request CANCEL to call
2009/4/1 Shaun Wingrin > Hi, > > Why does this warning occur and what are the implications of it? I'm > concerned about calls never getting hung up.! > > chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to > call '2f197e56611061a678c13b881b269...@411.2.139.106'. Giving up. > > Tx > Hello It's other end who is not aware if the call leg for that cancel, it is happening when some provider missconfigured the load balancing stuff for example, or call leg allready was destroyed for any reason. ___ -- 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] DAHDI with OSLEC
Marco Sambo wrote: > One thing! > I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel > 2.6.28 or newer to use oslec with DAHDI??? You don't need to, if you read me previous email you'll notice I'm running 2.6.27.19. Rebuild DAHDI with the instructions I linked to and you'll get the echo module with DADHI. It requires you download 2.6.28 but not that you are running 2.6.28. > > > > > > 2009/4/1 Marco Sambo > >> But I don't have also echo >> >> modinfo echo >> modinfo: could not find module echo >> >> >> >> >> >> 2009/4/1 Dave Fullerton >> >> Marco Sambo wrote: Mhmm. Thaht's strange! modinfo oslec --> modinfo: could not find module oslec and modinfo dahdi_echocan_oslec --> filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko license:GPL author: Tzafrir Cohen description:DAHDI OSLEC wrapper depends:dahdi vermagic: 2.6.26-1-486 mod_unload modversions 486 2009/3/31 Tzafrir Cohen > On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote: >> Hi, >> I've a problem: I can't configure DAHDI with ech canceller OSLEC. >> I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC. >> But when in /etc/dahdi/systems.conf I insert value > echocanceller=oslec,1-4, >> command dahdi_cfg - give me an error about oslec. > What is the output of: > > modinfo oslec > modinfo dahdi_echocan_oslec > > -- > Tzafrir Cohen > icq#16849755 > jabber:tzafrir.co...@xorcom.com >>> > > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > ___ >>> I'm not sure that is strange. When I build DAHDI with OSLEC I don't get >>> an oslec module, I get an echo module: >>> >>> r...@srvpbx:~# modinfo echo >>> filename: /lib/modules/2.6.27.19-smp/staging/echo/echo.ko >>> version:0.3.0 >>> description:Open Source Line Echo Canceller >>> author: David Rowe >>> license:GPL >>> srcversion: 285EC80D84DCE294A677160 >>> depends: >>> vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 >>> >>> r...@srvpbx:~# modinfo dahdi_echocan_oslec >>> filename: /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko >>> license:GPL >>> author: Tzafrir Cohen >>> description:DAHDI OSLEC wrapper >>> depends:dahdi,echo >>> vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 >>> >>> Try building DAHDI with the steps detailed here and see if you have >>> better luck: >>> >>> http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html >>> >>> -Dave >>> >>> ___ >>> -- 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 -- DO NOT SEND WITH THIS ACCOUNT ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Forking
Hello all, Probably a bad news for all... The Undercompetent Olle E Johansson decided to leave the asterisk team to create his own Voip server. The server will be called Minisk (due probably to his poor competence in Voip). Following that, Digium decides to stop any development on Asterisk and joined the Skype team to recreate a brand new paying licensed software. Any development will be halted for asterisk It seems that Mr Johansson has was poisoned by a fish in Brussels during the last Fosdem. We all hope qur Mr Johansson will quickly heal and return to the team of asterisk. Kind regards, Olivier Taylor ___ -- 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] DAHDI with OSLEC
One thing! I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel 2.6.28 or newer to use oslec with DAHDI??? 2009/4/1 Marco Sambo > But I don't have also echo > > modinfo echo > modinfo: could not find module echo > > > > > > 2009/4/1 Dave Fullerton > > Marco Sambo wrote: >> > Mhmm. Thaht's strange! >> > >> > modinfo oslec >> > --> >> > modinfo: could not find module oslec >> > >> > and >> > >> > modinfo dahdi_echocan_oslec >> > --> >> > filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko >> > license:GPL >> > author: Tzafrir Cohen >> > description:DAHDI OSLEC wrapper >> > depends:dahdi >> > vermagic: 2.6.26-1-486 mod_unload modversions 486 >> > >> > >> > >> > >> > >> > >> > 2009/3/31 Tzafrir Cohen >> > >> >> On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote: >> >>> Hi, >> >>> I've a problem: I can't configure DAHDI with ech canceller OSLEC. >> >>> I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC. >> >>> But when in /etc/dahdi/systems.conf I insert value >> >> echocanceller=oslec,1-4, >> >>> command dahdi_cfg - give me an error about oslec. >> >> What is the output of: >> >> >> >> modinfo oslec >> >> modinfo dahdi_echocan_oslec >> >> >> >> -- >> >> Tzafrir Cohen >> >> icq#16849755 >> >> jabber:tzafrir.co...@xorcom.com >> > >> >> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >> >> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir >> >> >> >> ___ >> >> I'm not sure that is strange. When I build DAHDI with OSLEC I don't get >> an oslec module, I get an echo module: >> >> r...@srvpbx:~# modinfo echo >> filename: /lib/modules/2.6.27.19-smp/staging/echo/echo.ko >> version:0.3.0 >> description:Open Source Line Echo Canceller >> author: David Rowe >> license:GPL >> srcversion: 285EC80D84DCE294A677160 >> depends: >> vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 >> >> r...@srvpbx:~# modinfo dahdi_echocan_oslec >> filename: /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko >> license:GPL >> author: Tzafrir Cohen >> description:DAHDI OSLEC wrapper >> depends:dahdi,echo >> vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 >> >> Try building DAHDI with the steps detailed here and see if you have >> better luck: >> >> http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html >> >> -Dave >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote host can't match request CANCEL to call
Hi, Why does this warning occur and what are the implications of it? I'm concerned about calls never getting hung up.! chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to call '2f197e56611061a678c13b881b269...@411.2.139.106'. Giving up. Tx___ -- 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] DAHDI with OSLEC
But I don't have also echo modinfo echo modinfo: could not find module echo 2009/4/1 Dave Fullerton > Marco Sambo wrote: > > Mhmm. Thaht's strange! > > > > modinfo oslec > > --> > > modinfo: could not find module oslec > > > > and > > > > modinfo dahdi_echocan_oslec > > --> > > filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko > > license:GPL > > author: Tzafrir Cohen > > description:DAHDI OSLEC wrapper > > depends:dahdi > > vermagic: 2.6.26-1-486 mod_unload modversions 486 > > > > > > > > > > > > > > 2009/3/31 Tzafrir Cohen > > > >> On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote: > >>> Hi, > >>> I've a problem: I can't configure DAHDI with ech canceller OSLEC. > >>> I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC. > >>> But when in /etc/dahdi/systems.conf I insert value > >> echocanceller=oslec,1-4, > >>> command dahdi_cfg - give me an error about oslec. > >> What is the output of: > >> > >> modinfo oslec > >> modinfo dahdi_echocan_oslec > >> > >> -- > >> Tzafrir Cohen > >> icq#16849755 > >> jabber:tzafrir.co...@xorcom.com > > > >> +972-50-7952406 mailto:tzafrir.co...@xorcom.com > >> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > >> > >> ___ > > I'm not sure that is strange. When I build DAHDI with OSLEC I don't get > an oslec module, I get an echo module: > > r...@srvpbx:~# modinfo echo > filename: /lib/modules/2.6.27.19-smp/staging/echo/echo.ko > version:0.3.0 > description:Open Source Line Echo Canceller > author: David Rowe > license:GPL > srcversion: 285EC80D84DCE294A677160 > depends: > vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 > > r...@srvpbx:~# modinfo dahdi_echocan_oslec > filename: /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko > license:GPL > author: Tzafrir Cohen > description:DAHDI OSLEC wrapper > depends:dahdi,echo > vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 > > Try building DAHDI with the steps detailed here and see if you have > better luck: > > http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html > > -Dave > > ___ > -- 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] DAHDI with OSLEC
Marco Sambo wrote: > Mhmm. Thaht's strange! > > modinfo oslec > --> > modinfo: could not find module oslec > > and > > modinfo dahdi_echocan_oslec > --> > filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko > license:GPL > author: Tzafrir Cohen > description:DAHDI OSLEC wrapper > depends:dahdi > vermagic: 2.6.26-1-486 mod_unload modversions 486 > > > > > > > 2009/3/31 Tzafrir Cohen > >> On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote: >>> Hi, >>> I've a problem: I can't configure DAHDI with ech canceller OSLEC. >>> I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC. >>> But when in /etc/dahdi/systems.conf I insert value >> echocanceller=oslec,1-4, >>> command dahdi_cfg - give me an error about oslec. >> What is the output of: >> >> modinfo oslec >> modinfo dahdi_echocan_oslec >> >> -- >> Tzafrir Cohen >> icq#16849755 >> jabber:tzafrir.co...@xorcom.com >> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir >> >> ___ I'm not sure that is strange. When I build DAHDI with OSLEC I don't get an oslec module, I get an echo module: r...@srvpbx:~# modinfo echo filename: /lib/modules/2.6.27.19-smp/staging/echo/echo.ko version:0.3.0 description:Open Source Line Echo Canceller author: David Rowe license:GPL srcversion: 285EC80D84DCE294A677160 depends: vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 r...@srvpbx:~# modinfo dahdi_echocan_oslec filename: /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko license:GPL author: Tzafrir Cohen description:DAHDI OSLEC wrapper depends:dahdi,echo vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 Try building DAHDI with the steps detailed here and see if you have better luck: http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html -Dave ___ -- 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] Queue data from within dialplan?
Many thanks - That is exactly what I want - I must have been using
poor search terms as I failed to find them on the Wiki previosuy :)
Regards,
Steve
2009/4/1 Lenz Emilitri :
> Are these functions what you are looking for?
>
> QUEUE_MEMBER_COUNT: Count number of members answering a queue
> QUEUE_MEMBER_LIST: Returns a list of interfaces on a queue
> QUEUE_WAITING_COUNT: Returns the number of callers currently waiting in a
> queue
>
> Just my two eurocents,
>
> l.
>
> 2009/3/31 Steve Davies
>>
>> Hi,
>>
>> It there any way of getting queue data from within a dialplan in order
>> to change call routing based on what is already happening? Something
>> like the following would be ideal:
>>
>> exten => X.,n,Set(WAITING=${QUEUE(qname|waiting)})
>> exten => X.,n,Set(TALKING=${QUEUE(qname|talking)})
>>
>> Can anyone suggest how I might achieve this?
>>
>> Thanks,
>> Steve
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Re: [asterisk-users] codec payload size
Hi, i got quiet the same problem, but with g711. Zoiper wan't really work if you got an ISDN Call, so "Zoiper" told me that the Asterisk send 16ms packets to zoiper and he can't handle 16ms. so if have to set 20ms, so what and how can i do this? Thx Timm - CPBX Austria by TMS IT-Dienst Hinterstadt 2 4840 Vöcklabruck T: (0720) 50 10 78 (Per ENUM kostenlos erreichbar) M: (0664) 479 79 25 F: (0720) 50 10 78-57 SIP: 2112377 (Terrasip) 0720501078 (Nemox) 0720721226 (PlatinPlus) Meine Mails werden mit Kaspersky AntiVirus überprüft! ___ -- 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] codec payload size
ContactTel Business wrote: > People should use .020 ms sample rates for RTP as it's the standard. 0.030 > was i think the old SPA implementations which caused MR, Roboto kind of > grabling. > > > You should find a way to patch your sip core i assume, but dev's could tell > you where. > > We offer 0.020 , Telcos offer 0.020 , Hardware also. Only a few > implementations still do 0.030 > How does "use 20ms, because so many things are too broken to work with anything else" make it a standard? I thought the standard was an RFC that says any number of 10ms frames may go into a packet, but if there's a SID frame it has to be the last in the packet. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stress asterisk voicemail
Hi friends. Can you help me to use SIPP to stress my asterisk voicemail? I want to send my own recorded media file to the voicemail system. Thanks. -- Linux User Registered #232544 Jabber : p...@jabberes.org Ekiga : p...@ekiga.net GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
Nice one, Olle ! :) On Wed, Apr 1, 2009 at 9:18 AM, Olle E. Johansson wrote: > * NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND > VIDEO TO MICROBLOGGING! ___ -- 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] What is the one thing that polycom can do...
Karl, I echoed your comment about a one button hit exit anywhere in the menu, that is so lame, although you can fake it by lifting the handset or hitting menu twice. I think a serious ergonomic study of the entire Polycom Soundpoint menuing interface is needed. It appears that little thought was given as to how it works. It actually looks like they tried to *guarantee* at least two or more presses for every single thing you'll ever need to do. Ok, I'm wrong: DND is a one button toggle. My bad! Anyway, limiting the number of button presses ensures longer service life of the product. Hmmm. /r ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
On Wed, 2009-04-01 at 09:18 +0200, Olle E. Johansson wrote: > > For more information, please do not contact Digium sales. > > To be released: 2009-04-01 > Should say enough... ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
On Wed, 01 Apr 2009 21:01:28 you wrote: > 2009/4/1 Michael > > > haw haw haw... > > > > April Fools Day is over in this part of the world. > > Hey dont kill the magic ! :) April Fools Day ends at 12.00pm (mid day) here. It is now 9:07pm. ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
2009/4/1 Michael > haw haw haw... > > April Fools Day is over in this part of the world. > > Hey dont kill the magic ! :) ___ -- 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] Queue data from within dialplan?
Are these functions what you are looking for?
QUEUE_MEMBER_COUNT: Count number of members answering a queue
QUEUE_MEMBER_LIST: Returns a list of interfaces on a queue
QUEUE_WAITING_COUNT: Returns the number of callers currently waiting in a
queue
Just my two eurocents,
l.
2009/3/31 Steve Davies
> Hi,
>
> It there any way of getting queue data from within a dialplan in order
> to change call routing based on what is already happening? Something
> like the following would be ideal:
>
> exten => X.,n,Set(WAITING=${QUEUE(qname|waiting)})
> exten => X.,n,Set(TALKING=${QUEUE(qname|talking)})
>
> Can anyone suggest how I might achieve this?
>
> Thanks,
> Steve
>
> ___
> -- 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
>
--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
haw haw haw... April Fools Day is over in this part of the world. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND VIDEO TO MICROBLOGGING! In a surprising move, Digium in partnership with Edvina today released a new channel driver for Asterisk, chan_tweet. The driver connects seamlessly to several microblogging platforms, including Twitter, Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of this new module is to add audio and video capabilities to microblogging, making the popular microblogging networks a new platform for VoIP and IP realtime communication. - "I have seen that the microblogging solutions building on the social network infrastructure have had enourmous unexploited capabilities", says Mill Biller at Digium, "I've used it for a long time both personally and for the company and we realized early that by adding IAX2 support, we could now take these platforms one giant leap forward by adding realtime multimedia. I can now spend evenings chit-chatting in audio and HD-resolution video with all my audience around the world instead of sending short text messages. It's truly awsome!" Digium contracted Edvina in Sweden, a well-known company in the Asterisk community and long-term Digium business partner, to build this solution. Edvina has many years of experience in building large- scale IAX2 networks, as well as doing development on the IAX2 support in Asterisk. - "IAX2 recently was published in an IETF RFC and we're pushing it heavily in all VoIP forums." says Olle Johansson of Edvina, "We're hoping that the IAX2FORUM will get a lot of new members that are willing to adopt this technology for their intranets, microblogging services and VoIP infrastructures. In the coming month, we will present more information about new partners with more than 100K users that are going to switch from old technologies, like Hype, SIP and H. 323. All of these protocols failed, either because they where proprietary or simply became too complex. SIP currently has more than 5.000 pages of documents describing all the features of the protocol and there's no single implementation of all of these to test with. Considering the protocol being over 10 years old, this is a sad story." - "We've done our best to fix the Asterisk SIP channel support for customers, but the customer base has been shrinking as more and more converted their networks to IAX2 and now, there's simply no one interested in us doing that work. We've stated over and over again that the SIP channel in Asterisk is broken and no one can prove us right or wrong, because the protocol is just too complex." * The Microblogmedia platform -- The Microblogmedia(TM) platform, developed by Digium and Edvina, let's users use any microblogging network to set up multimedia sessions. By compressing an IAX2 call setup event in the microblog message, web browsers and clients will connect automatically peer-2-peer if possible, or through the MicroBlogMediaRelay network that supports seamless NAT and firewall traversal by using automatic IPv6 tunnels. Asterisk 1.6.3, released later this month, will support this feature in the IAX2, H.323 and maybe in the old SIP channel (that is now marked deprecated). There is work on adding this feature to ISDN calls, by using messages in the D-channel for tunneling the IAX2 call setup messages. Digium's VoxSwitch will support this feature in the next release, planned for q3 2009. * Ending the Hype project --- In the same press release, Sock Stevens, product manager at Digium finally acknowledged that the Hype channel driver that was launched at Astricon 2008 will not be released after all. - "We found only one partner to test interoperability with, and that's not enough to make sure the channel driver being compatible with the protocol. And the protocol wasn't published in any RFC at all, or any other document. So we finally gave up. We're now dedicating resources for the new chan_tweet project and enhancing presence support in our IAX2 solution. With the installed base of IAX2 and the new MicroBlogMedia platform, this will be an even more impressive solution, reaching millions of IAX2 users in the enterprise as well as public sector and homes." * Technichal factoids - chan_tweet is the result of the project labelled "Codename orangepeel" amongst the development team and builds on the new "Pinemango" architecture. This is the first channel driver not connecting directly to the Asterisk core, but to the Pinemango API over Adversion, the Ruby framework developed by Phil Jaysip. - The MicroBlogMediaRelay IAX2 platform is an open distributed network that builds on IPv6 and a facebook application, thus using the enormous bandwidth provided for free by the Facebook(TM) platform - chan_tweet will be released with the core module in Open Source, but wi
