Re: [asterisk-users] MSet()
Anthony Messina schrieb: On Friday 12 December 2008 15:41:54 Mark Michelson wrote: would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. what about Set(ARRAY(var1,var2)=value1,value2) ? is the MSet() app a better/quicker way to do this? Just a word of warning: 1.4: Set(ARRAY(var1|var2)=val1\,val2) or: Set(ARRAY(var1,var2)=val1\,val2) 1.6: Set(ARRAY(var1,var2)=val1,val2) ? At least in 1.4 the , has to be escaped because otherwise you'd have a second argument to Set(). I still think that a plain Set(var1=val1); Set(var2=val2); is the most portable way which is least likely to cause any problems. portable = better Maybe not quicker. (Didn't do any benchmarks.) But certainly easier to read/write/maintain. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] docs for rxfax in 1.4 or app_fax in 1.6?
sean darcy schrieb:
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
TxFax() / RxFax() are available as a patch for 1.4. Somewhere.
Sorry, no pointers. But certainly easy to find.
1.6 comes with SendFax() / ReceiveFax() which use SpanDSP as
well and replace txfax/rxfax.
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
___
-- 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] SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
jonathan augenstine schrieb: Have you checked out OpenSBC (www.voip-info.org/wiki/view/*OpenSBC)?* http://www.voip-info.org/wiki/view/OpenSBC Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] ring back tone
Andrew Joakimsen schrieb: On Fri, Dec 12, 2008 at 18:57, Eric ManxPower Wieling e...@fnords.org wrote: Philipp Kempgen wrote: michel freiha schrieb: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? Dial(,,r) ? Much like violence and herding of llamas, the r option to Dial (and the Ringing app) almost never solve the problem they are intended to solve and frequently cause more, usually unforeseen, problems. Just say No! to r. If these are inbound calls you are answering, r can be acceptable. But ONLY on the final leg where the call might actually be ringing. But for outbound I fully agree. Let the carrier generate the ringing. r is never acceptable. :-) Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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 / Hylafax
On the subject of faxing - is there a way to interface Asterisk and Hylafax running on the same box? I have tried the directions at- http://www.voipinfo.org/wiki/view/T38modem+configuration+with+Asterisk and I just can't get it to work. Does anyone know of a way? (Even mention paid software)... Michael ___ -- 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 / Hylafax
On Sat, 13 Dec 2008 21:52:25 you wrote: Check out my tutorial at: http://blog.evaristesys.com/?p=24 It does use IAXmodem for conventional G.711u analog pass-thru (assuming you're using VoIP) rather than t38modem, but if you need to use T.38 I'm sure it can be easily adapted. 1. Do I need T.38 between Asterisk and Hylafax as I am using T.38 over SIP to the upline PSTN termination provider? 2. If this is workable, does using IAX to SIP cause extra CPU load / loss / issues? Totally off topic, but people may feel sorry for the residents of the Cook Islands. Their local ISP is selling 56 to 256k ADSL connections as 'broadband and the data caps and low and prices huge. http://www.oyster.net.ck/about/index.php?about=idsl (NZ$ 1.00 = US$0.55) Thanks, Michael ___ -- 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 / Hylafax
Michael schrieb: On the subject of faxing - is there a way to interface Asterisk and Hylafax running on the same box? IAXmodem. http://www.the-asterisk-book.com/unstable/faxserver-mit-iaxmodem-und-hylafax.html Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] prepaid solution
Use A2billing http://www.asterisk2billing.org/ Complete solution for prepaid calling card also. Shariq On Fri, Dec 12, 2008 at 11:45 PM, David fire ddf...@gmail.com wrote: prepaid solution for what? 2008/12/12 BERGANZ François franc...@acropolistelecom.net Hello, I am looking for a good prepaid solution. What is the best ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Asterisk / Hylafax
I am working my way through it and this is the error message- [2008-12-13 22:27:00] Modem started [2008-12-13 22:27:00] Setting device = '/dev/ttyIAX0' [2008-12-13 22:27:00] Setting owner = 'uucp:uucp' [2008-12-13 22:27:00] Setting mode = '660' [2008-12-13 22:27:00] Setting port = 4570 [2008-12-13 22:27:00] Setting refresh = 60 [2008-12-13 22:27:00] Setting server = '127.0.0.1' [2008-12-13 22:27:00] Setting peername = 'USERNAME' [2008-12-13 22:27:00] Setting secret = 'PASSWORD' [2008-12-13 22:27:00] Setting cidname = 'Net Trust Fax' [2008-12-13 22:27:00] Setting cidnumber = '098391001' [2008-12-13 22:27:00] Setting codec = alaw [2008-12-13 22:27:00] Opened pty, slave device: /dev/pts/2 [2008-12-13 22:27:00] Created /dev/ttyIAX0 symbolic link [2008-12-13 22:27:00] Registration failed. My configuration in iax.conf is as follows: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw mailboxdetail=yes iaxcompat=yes jitterbuffer=yes [FaxDSP0] type=friend username=USERNAME secret=PASSWORD host=127.0.0.1 port=4570 disallow=all allow=alaw qualify=yes PS: Is 'peername' in the IAX config the same as the iax.conf 'username'? ___ -- 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 / Hylafax
For some odd reason the call registration issue doesn't seem to stop it working, except a few seconds after Hylafax answers the call it hangs up, I suspect because Asterisk only supports T38 pass through. Here is my data path PSTN = T.38 aware SIP provider = Internet = My machine (Asterisk) = IAX modem on localhost = Hylafax. While I am aware from being on the Hylafax mailing list that IAX does work, no one there has been able to advise of a way it can be done with a SIP upline provider. I think people generally use it for ISDN PRI = Asterisk = IAX modem = Hylafax setups. Now Call Weaver apparently does have some better fax capabilities though I can't find any info on how to integrate it with Hylafax. Does anyone know how it can be done with Call Weaver and/or if there is any 3rd party software to enable what I am after? Michael ___ -- 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] G729 codec files
Which of these files is best for a Core2 Duo? I'm not sure whether to choose Pentium or i686. http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/ ___ -- 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] PickupChan()
---cut--- [Description] PickupChan(channel[channel...]): This application can pickup any ringing channel ---cut--- Does channel really mean channel name (SIP/3000-abde1234) or rather technology/resource (SIP/3000)? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] docs for rxfax in 1.4 or app_fax in 1.6?
Hi,
I can not find a version that works with asterisk = 1.4.22.
Agx-addons provide a packages that I used successfully with version
below asterisk 1.4.22.
Loic
On Sat, 2008-12-13 at 09:37 +0100, Philipp Kempgen wrote:
sean darcy schrieb:
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
TxFax() / RxFax() are available as a patch for 1.4. Somewhere.
Sorry, no pointers. But certainly easy to find.
1.6 comes with SendFax() / ReceiveFax() which use SpanDSP as
well and replace txfax/rxfax.
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20 20
Fax: +352 20 90
ldide...@mixvoip.com
http://www.mixvoip.com
___
-- 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] docs for rxfax in 1.4 or app_fax in 1.6?
Hello,
which package are you using to get the application ReceiveFAX under
asterisk 1.4?
Loic
On Fri, 2008-12-12 at 17:37 -0600, Anthony Messina wrote:
On Friday 12 December 2008 12:08:55 sean darcy wrote:
I just want to pdf and email faxes coming in over pstn on a TDM400P.
Outgoing faxes would just go out over pstn, not through asterisk.
All the voipinfo , etc, howto's are quite complicated. And most use
third party apps like Hylafax.
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
i'm working on a email - fax gateway right now to do just that. it works
well, but is unpolished and basically undocumented at this point. you can
see
the work in svn here: http://messinet.com/viewvc/asterisk-fax-gw/trunk/
___
-- 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
--
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20 20
Fax: +352 20 90
ldide...@mixvoip.com
http://www.mixvoip.com
___
-- 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] docs for rxfax in 1.4 or app_fax in 1.6?
On Saturday 13 December 2008 11:34:07 Loic Didelot wrote: Hello, which package are you using to get the application ReceiveFAX under asterisk 1.4? since this is still in development, for me, i'm working with asterisk 1.6. this package is mostly independent of the fax system within asterisk (just change the dialplan to use tx_fax instead of sendfax, basically. as far as getting app_txfax and app_rxfax to work in 1.4, i never tried that. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Please explain the meaning of the output of lsmod | grep ztdummy?
lsmod | grep ztdummy ztdummy38856 0 zaptel231496 3 ztdummy ___ -- 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] Please explain the meaning of the output of lsmod | grep ztdummy?
On 20:30, Sat 13 Dec 08, Shaun Wingrin wrote: lsmod | grep ztdummy ztdummy38856 0 zaptel231496 3 ztdummy it means you have ztdummy loaded so you can do Meetme rooms and IAX2 trunking without actual zaptel hardware. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] MSet()
On Saturday 13 December 2008 02:32:01 Philipp Kempgen wrote: Anthony Messina schrieb: On Friday 12 December 2008 15:41:54 Mark Michelson wrote: would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. what about Set(ARRAY(var1,var2)=value1,value2) ? is the MSet() app a better/quicker way to do this? Just a word of warning: 1.4: Set(ARRAY(var1|var2)=val1\,val2) or: Set(ARRAY(var1,var2)=val1\,val2) 1.6: Set(ARRAY(var1,var2)=val1,val2) ? At least in 1.4 the , has to be escaped because otherwise you'd have a second argument to Set(). I still think that a plain Set(var1=val1); Set(var2=val2); is the most portable way which is least likely to cause any problems. portable = better Maybe not quicker. (Didn't do any benchmarks.) But certainly easier to read/write/maintain. While ARRAY() certainly would work this way for setting multiple variables at once, the intended usage is for setting multiple variables from the result of a command which generates multiple answers (read: multiple points of data). The usual case for this is func_odbc. -- Tilghman ___ -- 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 video support
Dear all, does anybody try Asterisk 1.4 with video support and using canreinvite=no in sip.conf ??? Because I need Asterisk to be in the middle of voice and video calls among end users. Partucularly I will use H.264 video codec. Is it possible ??? Thanks s lot Alejandro Yahoo! Cocina Recetas prácticas y comida saludable http://ar.mujer.yahoo.com/cocina/___ -- 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] prepaid solution
hı I mustafa. a2billing IF YOU WANT AND I CAN SET Date: Fri, 12 Dec 2008 16:45:26 -0200From: ddf...@gmail.comto: asterisk-us...@lists.digium.comsubject: Re: [asterisk-users] prepaid solutionprepaid solution for what? 2008/12/12 BERGANZ François franc...@acropolistelecom.net Hello, I am looking for a good prepaid solution. What is the best ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___-- Bandwidth and Colocation Provided by http://www.api-digital.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. _ Windows Live Messenger'ın için Ücretsiz 30 İfadeyi yükle http://www.livemessenger-emoticons.com/funfamily/tr-tr/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, Dec 13, 2008 at 04:10:47PM +1300, Michael wrote: Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. Here is some info for the UK :- http://www.ofcom.org.uk/telecoms/ioi/numbers/numbers_administered/ I've spotted a small number of errors in the data, and I can give you the details if you want. Cheers Ben ___ -- 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 video support
On Sat, Dec 13, 2008 at 19:35, Alejandro Facultad alejandro_facul...@yahoo.com.ar wrote: Dear all, does anybody try Asterisk 1.4 with video support and using canreinvite=no in sip.conf ??? Because I need Asterisk to be in the middle of voice and video calls among end users. Partucularly I will use H.264 video codec. Is it possible ??? Yes, no problem Éric Thanks s lot Alejandro ¡Buscá desde tu celular! Yahoo! oneSEARCH ahora está en Claro http://ar.mobile.yahoo.com/onesearch ___ -- 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] docs for rxfax in 1.4 or app_fax in 1.6?
Philipp Kempgen wrote:
sean darcy schrieb:
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
TxFax() / RxFax() are available as a patch for 1.4. Somewhere.
Sorry, no pointers. But certainly easy to find.
1.6 comes with SendFax() / ReceiveFax() which use SpanDSP as
well and replace txfax/rxfax.
Philipp Kempgen
Thanks for the reply.
I'll probably try ReceiveFax() if it's in 1.6. Patching outside apps
sometimes leads to version trouble.
Any docs for ReceiveFax()? Or, if I'm really lucky, an example?
sean
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Dec 12, 2008, at 7:10 PM, Michael wrote: Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. [snip] I'll avoid the good discussion on this thread that has been made already to date. The answer is: It's not simple. I wish it was, but that data is typically in the hands of companies who do NOT want to share it, because errors on your part make money on their part, and those companies don't really want to see you exist at all in the first place. It would be great if this could be a shared resource of some sort, but I don't expect that we'll ever see that with E.164 numbering - the entrenched interests in that number space have zero interest in making the data available for many political, technical, and fiscal reasons. In the past, I've received a full breakdown of rates from my carriers for North America, as I have typically purchased directly from North American carriers who give me a 5, 6, or 7 tier model of prices across the country. LNP has confused the issue a bit, but not enough to warrant dips into the LNP database to determine if mobile numbers are being delivered on landline, or vice versa. For most of the world not in country code 1, I typically get just two rates: landline and mobile. In some very large nations, mobile may be broken down into a few carriers, but not often. Carriers almost always will NOT give out number ranges for what is mobile versus what is landline. They just say Mobile: $V.WXYZ Landline: $A.BCDE and leave it to me to figure out what is what. This, of course, is disingenuous - obviously they know what is mobile and what is landline, since they need it for their own billing purposes. But they would NEVER try to help out a customer like that, since it makes the customer more informed and less likely to just swallow an incorrect billing. So in a recent job (not Digium) I had to get the Telcordia GDDS database. Cost varies depending on what kind of use you have, but I seem to recall ranging from $20k per year to $Absurd per year. It was useful - recall that mobile rates can be 10x or even 20x what landline rates are, so billing your customers in the same bracket as you are billed by your vendors becomes startlingly important when you hit the millions of minutes per month mark in international termination. http://www.telcordia.com/products_services/trainfo/catalog_details.html http://www.trainfo.com/products_services/tra/downloads/gdds.pdf Then, you'll need to create a database that tracks what each one of your vendors gives you for mappings, create a linkage between their terminology or number formats to the GDDS database, and extract to create a routing table. Non-trivial, but not super-difficult. Get a big machine with gobs of RAM and store all of your databases in cache. The US I seem to recall had (for each carrier) something like 420,000 rows in one configuration. Multiply by the number of carriers you are using, add in the 60,000 or so rows for international per carrier... it gets to be a pretty plump routing table. You must be _this_ tall to ride this wholesale ride. We created an external Java app that would take the origin number out of our Asterisk arrays and then do a database lookup. Total delay between receipt of the INVITE and the INVITE being out to the best vendor was around .3 seconds, so not bad. But there is a reason that there are whole companies doing routing and rating engines. JT --- 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/ ___ -- 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] FBI issues VoIP security warning on Asterisk --which version?
On Dec 12, 2008, at 4:51 AM, Khaled Chehab wrote: Dear All FBI issues VoIP security warning on Asterisk -- but which version? Any one know which version ? Regards The short answer to your question is 1.2 and 1.4 The longer answer is Read this security release from March of this year, which is what the FBI was talking about: http://downloads.digium.com/pub/security/AST-2008-003.pdf The even longer, but more complete answer is: http://blogs.digium.com/2008/12/06/sip-security-and-asterisk/ JT --- 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/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sun, 14 Dec 2008 19:57:20 you wrote: On Dec 12, 2008, at 7:10 PM, Michael wrote: Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. [snip] I'll avoid the good discussion on this thread that has been made already to date. The answer is: It's not simple. I wish it was, but that data is typically in the hands of companies who do NOT want to share it, because errors on your part make money on their part, and those companies don't really want to see you exist at all in the first place. It would be great if this could be a shared resource of some sort, but I don't expect that we'll ever see that with E.164 numbering - the entrenched interests in that number space have zero interest in making the data available for many political, technical, and fiscal reasons. [snip] Thanks John. You are clearly someone who has experience, and is well informed, though your contribution reflects those of other people from USA, whereas from where I (and most others) are from, USA is only 1 or 2 billing destinations out of 100+. So the validity of the idea/suggested project still stands. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, Dec 13, 2008 at 10:57:20PM -0800, John Todd wrote: On Dec 12, 2008, at 7:10 PM, Michael wrote: Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. [snip] I'll avoid the good discussion on this thread that has been made already to date. The answer is: It's not simple. Right. So for those of us who want to do simple things and avoid complicated stuff such as telephony in shoddy continent of North America, could you please provide data for your country? So far we have AU, IL and NZ. -- 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] Country numbering plan resources
On Sun, 14 Dec 2008 20:14:22 Tzafrir Cohen wrote: On Sat, Dec 13, 2008 at 10:57:20PM -0800, John Todd wrote: On Dec 12, 2008, at 7:10 PM, Michael wrote: Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. [snip] I'll avoid the good discussion on this thread that has been made already to date. The answer is: It's not simple. Right. So for those of us who want to do simple things and avoid complicated stuff such as telephony in shoddy continent of North America, could you please provide data for your country? So far we have AU, IL and NZ. We have Uruguay as well. ___ -- 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
