Re: [Asterisk-Users] Headless Linux system for Asterisk
Hi, >- Original Message - >From: "Jeremy McNamara" <[EMAIL PROTECTED]> >Subject: Re: [Asterisk-Users] Headless Linux system for Asterisk > > Just make sure you make sure the BIOS is set not to halt the system on > any errors. What about make it work without a graphic card too? If the MB has the graphic on board, usually has just 2 or 3 PCI slots which is not acceptable for an * box. They are many new mainboards which does support only AGP 4x and 8x. Why to pay for a graphic card from the new generation if you don't need it... BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] after hours
Sorry what did you say you had in your hand ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, 19 December 2003 2:34 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] after hours On Thu, 2003-12-18 at 15:51, matt wrote: > Although it's hard to see the original proverb writer saying RTFM Not really. The manual contains the teachings necessary to learn for yourself. So if nudging yourself down the road to self enlightenment is nothing more than reminding you there are fish in the stream, and a rod in your hand. > Matt > > Andrew Thompson wrote: > > "Give a man a fish and he eats for a day. Teach him to fish and he > eats for a lifetime." > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Telemarketer Torture
Cees de Groot schrieb: > > Andrew Thompson <[EMAIL PROTECTED]> said: > >While an exceptionally devious concept, I don't think it'd work out like you > >planned. Wouldn't that mean you'd have to dial out the 900 number yourself, > >meaning You would be charged for the 900 call. > > > At least with ISDN, you can deflect the call AFAIK. Dunnow how this > works out billing-wise, though, never used it. with german ISDN providers the forwarded part of the connection is charged to your account. The as if you would forward it yourself. The only difference ist "used lines". stiffen up your wellliked marketeer with a chrismas-tree! uwe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Headless Linux system for Asterisk
Dan <[EMAIL PROTECTED]> said: >Why to pay for a graphic card from the new generation if you don't need >it... > Or you buy a decent A brand pizzabox, which has everything on-board. We use IBM xSeries, which has everything you want built-in and comes with management hardware so I can take over the console remotely with a telnet session (including the whole boot sequence, power on/off, etcetera). Personally, I like to have a graphics card in my headless servers, sometimes the easiest way to work with a box is to plug in a console and monitor. In the datacenter, the xSeries 1U boxes are all looped together with special cables and share a single KVM port (really nice); at home, my headless servers are all connected to a cheapo Belkin KVM switch, works great and saves a lot of hassle with disconnected keyboards, serial console fiddling, etcetera. -- Cees de Groot http://www.tric.nl <[EMAIL PROTECTED]> tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for the telephone. Something that would combine the functionality of a (adsl modem+) router and a SIP telephone adaptor in one box. I would appreciate any info that you might have on this. regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interconnecting Panasonic KX-TD1232 digital PBX and *
Hi all, There is someone with some experience interconnecting a Panasonic digital PBX (KX-TD1232) with Asterisk? Thank you and best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP call waiting disable?
Hello, I`m using the Dlink DG-104s with asterisk, it works ok for incoming outgoing calls. The problem is, when for exmple the line on the dg104s is off hook and I dial that extension I dont get a busy but a ringing tone. And no call waiting signals on the dg104s side. Is there a way to detec busy, and to disable call waiting? thanks -- Anton Yurchenko<[EMAIL PROTECTED]> Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is/isn't affected by "reload"?
> help reload says: > Reloads configuration files for all modules which support reloading. > > Well that's all fine and dandy, but which modules actually support > reloading? MGCP does not support reloading Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
Hi! > I was wondering whether any of you have experience/info on Cable and/or ADSL > modems that would come together with a SIP phone adaptor. What I am > interested in is something that would plug directly into you ISP's cable (be > it ethernet or adsl/phoneline), would combine a modem/router/nat such that > on the other you could simply plug in your RJ-45 cable for your PC and a > RJ-11 cable for the telephone. Something that would combine the > functionality of a (adsl modem+) router and a SIP telephone adaptor in one > box. This is not exactly what you are asking for, but its getting close: http://www.voip-info.org/tiki-index.php?page=VOIP+Phones Look at the "Clipcomm" devices (Korea). I'd be interested in any reports & experiences. Next to that the newer Grandstream firmware now supports PPPoE (but now routing or port forwarding). Not sure what the CISCO devices offer in this respect. I wouldn't want to have ADSL modem & router coupled in one device, that'll make your router useless if you move to cable modem etc. So better not integrate too many task into one piece of hardware. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to H.323 without gatekeeper
I've read through the archives and have picked up that * does not need a gatekeeper to talk directly with an H323 handset to send and receive calls. I'm trying to go PSTN*-H323 and all the examples that I can find use a gatekeeper. Are there any examples or hints for doing it without the gatekeeper? many thanks in advance Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sphinx
Hi,
I'm, doing the same, but with a php agi, and invoking a modified script to use
sphinx:
/usr/local/bin/decoder2
#!/bin/sh
BASE=6403
S2BATCH=sphinx2-continuous
HMM=/usr/local/share/sphinx2/model/hmm/6k
TASK=/usr/local/share/sphinx2/model/lm/${BASE}
CTLFILE=/usr/local/share/sphinx2/model/lm/${BASE}/${BASE}.ctl
#BASE=$1
$S2BATCH -nbest 1 -nbestdir /tmp -verbose 0 -adcin TRUE -adcext wav -ctlfn
${CTLFILE} -ctloffset 0 -ctlcount 1 -datadir ${TASK} -samp 8000
-agcmax TRUE -langwt 6.5 -fwdflatlw 8.5 -rescorelw 9.5 -ugwt 0.5 -fillpen
1e-10 -silpen 0.005 -inspen 0.65 -top 1 -topsenfrm 3 -topsenthresh -7
-beam 2e-06 -npbeam 2e-06 -lpbeam 2e-05 -lponlybeam 0.0005 -nwbeam 0.0005
-fwdflat FALSE -fwdflatbeam 1e-08 -fwdflatnwbeam 0.0003 -bestpath TRUE
-kbdumpdir ${TASK} -lmfn ${TASK}/${BASE}.lm -dictfn ${TASK}/${BASE}.dic
-noisedict ${HMM}/noisedict -phnfn ${HMM}/phone -mapfn ${HMM}/map -hmmdir
${HMM} -hmmdirlist ${HMM} -8bsen TRUE -sendumpfn ${HMM}/sendump -cbdir ${HMM}
My php-test.agi:
function __write__($line)
{
print $line."\n";
}
function __read__()
{
global $in;
$input=str_replace("\n","",fgets($in,4096));
return $input;
}
function festival($texto)
{
__write__("EXEC FESTIVAL \"$texto\"");
return __read__();
}
function record($filename, $digits)
{
__write__("RECORD FILE $filename wav $digits 15000 BEEP");
return __read__();
}
$in=fopen("php://stdin","r");
festival("ingrese su mensaje");
record("mensaje","#");
$comando="/usr/local/bin/decoder2";
$value=file("/tmp/mensaje.hyp");
$value=$value[0];
$numero=trim(substr($value,strpos($value,">")+1));
if($numero!="")
festival("usted a seleccionado el numero $numero");
exec($comando);
El Jueves 18 Diciembre 2003 17:40, Kevin Bockman escribió:
> Hi. I just started trying to play with Sphinx. I followed their site as
> far as running sphinx-server. It is listening on the default port. I
> copied sphinx2-simple to another file and changed sphinx2-continuous to
> sphinx2-server.
>
> So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to
> do? Here's what I get:
>
> debian:~# sphinx2-simple2
>
> sphinx2-simple:
> Demo CMU Sphinx2 decoder called with command line arguments.
>
>
> ioctl(SETDUPLEX) failed: Invalid argument
> Calibrating background noise level...done
> server.c(443): Bad or missing port# argument, using 7027
> srvcore.c(382): Listening at port 7027
> srvcore.c(409): Connected 192.168.1.99 at Thu Dec 18 15:24:19 2003
>
> Hit to start listening, q to quit client connection
>
> -- Executing Answer("SIP/test-ff55", "") in new stack
> -- Executing EAGI("SIP/test-ff55", "eagi-sphinx-test") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/eagi-sphinx-test
> Environment: 'agi_request' is 'eagi-sphinx-test'
> Environment: 'agi_channel' is 'SIP/test-ff55'
> Environment: 'agi_language' is 'en'
> Environment: 'agi_type' is 'sip'
> Environment: 'agi_uniqueid' is '1071786651.21'
> Environment: 'agi_callerid' is '"blah" <1234>'
> Environment: 'agi_dnid' is 'unknown'
> Environment: 'agi_rdnis' is 'unknown'
> Environment: 'agi_context' is 'default'
> Environment: 'agi_extension' is '911'
> Environment: 'agi_priority' is '2'
> Environment: 'agi_enhanced' is '1.0'
> Environment: 'agi_accountcode' is ''
> Ooh, got a response from Asterisk: '200 result=0 endpos=46560'
> 1. Result is '200 result=0 endpos=46560'
> Ooh, got a response from Asterisk: '200 result=0 endpos=30720'
> 2. Result is '200 result=0 endpos=30720'
> -- Playing 'digits/20' (language 'en')
> -- Playing 'digits/3' (language 'en')
> -- Playing 'digits/million' (language 'en')
> -- Playing 'digits/4' (language 'en')
> -- Playing 'digits/hundred' (language 'en')
> -- Playing 'digits/50' (language 'en')
> -- Playing 'digits/2' (language 'en')
> -- Playing 'digits/thousand' (language 'en')
> -- Playing 'digits/3' (language 'en')
> -- Playing 'digits/hundred' (language 'en')
> -- Playing 'digits/40' (language 'en')
> -- Playing 'digits/5' (language 'en')
> Ooh, got a response from Asterisk: '200 result=0'
> 3. Result is '200 result=0'
> -- Playing 'demo-enterkeywords' (language 'en')
> Ooh, got a response from Asterisk: '200 result= (timeout)'
> 4. Result is '200 result= (timeout)'
> Ooh, got a response from Asterisk: '200 result=0 endpos=9440'
> 5. Result is '200 result=0 endpos=9440'
> -- AGI Script eagi-sphinx-test completed, returning 0
>
> Is the endpos number something significant? What is it referring to?
>
> Am I doing this right?
>
> Does anyone have any other EAGI Sphinx examples? Maybe something that asks
> for you to say a word and it puts it into text?
>
> Sorry, this is pretty neat but there's hardly any information on it. I've
> tried searching the lists and I see a couple people that have used it but
> that's all.
>
> Thanks,
>
> Kevin B.
>
> ___
Re: [Asterisk-Users] Re: Expressions - solved (chan_local)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 19 December 2003 04:18, John Todd wrote: > An additional modifier to Dial was added specifically for Local > channel types. If you add the "/n" modifier at the end of a > chan_local call, then the variables will be erased upon passage > through the system. Otherwise, you'll get the variables from the > first leg of the call passed through upon native bridge - there is a > brief interval, during the "ringing" phase of the second leg, where > the second leg has it's own variables, but then those get wiped out > upon bridging. Adding the "/n" option prevents this. This certainly solved one half of my problem! Thanks! - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/4v2V2TEAILET3McRAiQEAJ9NfSaDHU/VK88y6sAFo8dZU79XuwCeNziR gtryLzCGVQ21i75RVoKIk8Y= =sA1O -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
bam wrote:
I've read through the archives and have picked up that * does not need
a gatekeeper to talk directly with an H323 handset to send and receive
calls.
I'm trying to go PSTN*-H323 and all the examples that I can
find use a gatekeeper. Are there any examples or hints for doing it
without the gatekeeper?
many thanks in advance
Brian
[your_context]
exten => _9XX,1,Dial,H323/78632${EXTEN:[EMAIL PROTECTED]|30
exten => _9XX,2,Busy
exten => _9XX,102,Busy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
--
-
Best Regards,
Pavel Litvinenko.
ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi & Eicon Diva problem
Hi Patrick, I have exactly the same problem with diva BRI-2M. Did you solve it? Regards, Daniel Patrick a écrit: Hello, I have an issue getting the chan_capi module to load in asterisk cvs from today. Plain 2.4.20 kernel with melware patches for the Eicon Diva Server Bri card. I load the modules with: modprobe -v divas divacapi I load the firmware with: divactrl load -c 1 -f ETSI -vd6 Output in /var/log/messages is: Nov 9 19:26:26 voice kernel: Eicon DIVA - DIDD table (http://www.melware.net) Nov 9 19:26:26 voice kernel: divadidd: Rel:2.0 Rev:1.13 Build:102-51(local) Nov 9 19:26:26 voice kernel: Eicon DIVA Server driver (http://www.melware.net) Nov 9 19:26:26 voice kernel: divas: Rel:2.0 Rev:1.45 Build: 102-52(local) Nov 9 19:26:26 voice kernel: divas: support for: BRI/PCI PRI/PCI adapters Nov 9 19:26:26 voice kernel: divas: Diva Server BRI-2M PCI bus: fn: 0048 insertion. Nov 9 19:26:26 voice kernel: PCI: Found IRQ 14 for device 00:09.0 Nov 9 19:26:26 voice kernel: PCI: Sharing IRQ 14 with 00:04.2 Nov 9 19:26:26 voice kernel: PCI: Sharing IRQ 14 with 00:06.0 Nov 9 19:26:26 voice kernel: divas: bus: fn: 0048 fix latency. Nov 9 19:26:26 voice kernel: divas: Diva Server BRI-2M PCI IRQ:14 SerNo:1884 Nov 9 19:26:26 voice kernel: divas: started with major 254 Nov 9 19:26:26 voice kernel: divas: thread started with pid 18151 chan_capi-0.3.0 is properly installed. When I start asterisk I get in /var/log/messages: Nov 9 19:27:21 voice modprobe: modprobe: Can't locate module char-major-68 And the asterisk console says: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found NOTICE[16384]: File chan_capi.c, Line 2646 (load_module): CAPI not installed! WARNING[16384]: File loader.c, Line 305 (ast_load_resource): chan_capi.so: load_module failed, returning -1 WARNING[16384]: File chan_capi.c, Line 2733 (unload_module): Unable to unregister from CAPI! WARNING[16384]: File loader.c, Line 351 (load_modules): Loading module chan_capi.so failed! I have /dev/capi20 and /dev/capi20.00-19 devices with major 68. But I also have /dev/Divas with major 254 crw---1 root root 68, 0 Jan 30 2003 capi20 crw-r--r--1 root root 254, 0 Nov 9 19:26 Divas Any ideas? TIA, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 question
My understanding is that if the codec type matches on each leg of the call then Asterisk is using "pass through". The PBX simply has to relay the RTP packets unchanged from one device to another. If you are going from a phone using codec tupe A to codec type B, Asterisk must perform a conversion which requires support for both codecs. iTS [EMAIL PROTECTED] wrote: Hi, where can I find info on configuring pass-through mode. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of SW Sent: Jueves, 18 de Diciembre de 2003 11:29 p.m. To: asterisk users Cc: Clif Jones Subject: [Asterisk-Users] G729 question Hi Clif, My experience with G.729 and asterisk is not good. My first registration was good, it worked. Then I bought more license and tried to upgrade it, it blew everything off. Still waiting Digium support to give me a helping hand. If you use pass-through feature then I guess you are fine. I have SIP users going to h.323 g/w and I need g.729. So now I have it in pass-through mode, I think that requires less CPU overhead and I do not have to mess with licenses. Cheers SW Message: 5 Date: Thu, 18 Dec 2003 13:59:06 -0500 From: Clif Jones <[EMAIL PROTECTED]> To: asterisk users <[EMAIL PROTECTED]> Subject: [Asterisk-Users] G729 question Reply-To: [EMAIL PROTECTED] I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this answer. I know that G729A is low complexity which seems to be what Cisco 7960's use but I have some others that support G729B which has comfort noise and reduced transmission during silence. If anyone knows how the different G729 codecs interoperate I would be eager to know. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.551 / Virus Database: 343 - Release Date: 11/12/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.551 / Virus Database: 343 - Release Date: 11/12/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
- Original Message - From: "Dawid Mielnik" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, December 19, 2003 6:01 AM Subject: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem) > > Hi all, > > I was wondering whether any of you have experience/info on Cable and/or ADSL > modems that would come together with a SIP phone adaptor. What I am > interested in is something that would plug directly into you ISP's cable (be > it ethernet or adsl/phoneline), would combine a modem/router/nat such that > on the other you could simply plug in your RJ-45 cable for your PC and a > RJ-11 cable for the telephone. Something that would combine the > functionality of a (adsl modem+) router and a SIP telephone adaptor in one > box. > > I would appreciate any info that you might have on this. > The only thing that I can think of that might come close is a Cisco 2600 series. They are modular routers with two or three expansion ports. You can for sure get an Ethernet connection as one of the modules, but I'm not sure about ADSL or Cable. They have ports for VOIP, but I'm not familiar with them: http://www.cisco.com/en/US/products/hw/routers/ps259/products_relevant_inter faces_and_modules.html (Scroll about half way down the page, you'll see several sections of VOIP stuff.) - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] x100P incoming
SW <> wrote: > Hi Gurus Not a guru, but I'll see what I can do. > How do I make x100P does not answer incoming calls ? The only thing that springs to mind is that you create an incoming context, and have an extension like: Exten => s,1,Wait(1000) Dunno if it will work or not, but that's the only thing that springs to mind. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom phones update
On Thu, 2003-12-18 at 17:09, Juan J. Sierralta P. wrote: > Does anybody know if Polycoms has a "three finger salute" as Cisco 79XX > does ? I really hate to unplug ethernet cable since you have to release > the stand first. I respond myself, hold down: Volume+, Volume-, Hold and Messages. > -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi & Eicon Diva problem
I ahve found the pb: there was some /dev/... device mising. Regards, Daniel ANDRE Daniel ANDRE a écrit: Hi Patrick, I have exactly the same problem with diva BRI-2M. Did you solve it? Regards, Daniel Patrick a écrit: Hello, I have an issue getting the chan_capi module to load in asterisk cvs from today. Plain 2.4.20 kernel with melware patches for the Eicon Diva Server Bri card. I load the modules with: modprobe -v divas divacapi I load the firmware with: divactrl load -c 1 -f ETSI -vd6 Output in /var/log/messages is: Nov 9 19:26:26 voice kernel: Eicon DIVA - DIDD table (http://www.melware.net) Nov 9 19:26:26 voice kernel: divadidd: Rel:2.0 Rev:1.13 Build:102-51(local) Nov 9 19:26:26 voice kernel: Eicon DIVA Server driver (http://www.melware.net) Nov 9 19:26:26 voice kernel: divas: Rel:2.0 Rev:1.45 Build: 102-52(local) Nov 9 19:26:26 voice kernel: divas: support for: BRI/PCI PRI/PCI adapters Nov 9 19:26:26 voice kernel: divas: Diva Server BRI-2M PCI bus: fn: 0048 insertion. Nov 9 19:26:26 voice kernel: PCI: Found IRQ 14 for device 00:09.0 Nov 9 19:26:26 voice kernel: PCI: Sharing IRQ 14 with 00:04.2 Nov 9 19:26:26 voice kernel: PCI: Sharing IRQ 14 with 00:06.0 Nov 9 19:26:26 voice kernel: divas: bus: fn: 0048 fix latency. Nov 9 19:26:26 voice kernel: divas: Diva Server BRI-2M PCI IRQ:14 SerNo:1884 Nov 9 19:26:26 voice kernel: divas: started with major 254 Nov 9 19:26:26 voice kernel: divas: thread started with pid 18151 chan_capi-0.3.0 is properly installed. When I start asterisk I get in /var/log/messages: Nov 9 19:27:21 voice modprobe: modprobe: Can't locate module char-major-68 And the asterisk console says: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found NOTICE[16384]: File chan_capi.c, Line 2646 (load_module): CAPI not installed! WARNING[16384]: File loader.c, Line 305 (ast_load_resource): chan_capi.so: load_module failed, returning -1 WARNING[16384]: File chan_capi.c, Line 2733 (unload_module): Unable to unregister from CAPI! WARNING[16384]: File loader.c, Line 351 (load_modules): Loading module chan_capi.so failed! I have /dev/capi20 and /dev/capi20.00-19 devices with major 68. But I also have /dev/Divas with major 254 crw---1 root root 68, 0 Jan 30 2003 capi20 crw-r--r--1 root root 254, 0 Nov 9 19:26 Divas Any ideas? TIA, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Huge traffic with iaxtel.com (without making calls)!!!
Hi all, My * box is registered with IAXTEL too. The problem is that iaxtel.com [69.73.19.178] makes me a lot of traffic (both inbound and outbound) on my external IP address (* is behind a NAT router and only UDP port 4569 is open to the server). More, all by "browsing and download" traffice is made through a local proxy Checkig the logs of my Internet connection, they are around 120MB/day both inbound and outbound traffic, even at some strange local hours, when nobody use the phone. I have discovered that because of the very low performance of my connection in the last period. I have checked the Asterisk calls log too and they are no calls at that hours. Because my cable connection is limited to 1GB of traffic per month (max between in and out) I am forced to stop the DIAX "CallMe" feature till this issue will be solved. If someone can explain this, please send me a direct mail. Sorry for the inconvenience. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi documentation
Hello, Is there any docmnetation on the configuration of chan_capi: syntax of capi.conf? Dial string configuration? Best Regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
- Original Message -
From: "Pavel Litvinenko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, December 19, 2003 8:42 AM
Subject: Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
> bam wrote:
>
> > I've read through the archives and have picked up that * does not need
> > a gatekeeper to talk directly with an H323 handset to send and receive
> > calls.
> >
> > I'm trying to go PSTN*-H323 and all the examples that I can
> > find use a gatekeeper. Are there any examples or hints for doing it
> > without the gatekeeper?
> >
> > many thanks in advance
> >
> > Brian
>
>
> [your_context]
>
> exten => _9XX,1,Dial,H323/78632${EXTEN:[EMAIL PROTECTED]|30
> exten => _9XX,2,Busy
> exten => _9XX,102,Busy
>
What's the 78632? Is that something you have to dial, like country/area code
+ 6 digits?
-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.
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Re: [Asterisk-Users] chan_capi & Eicon Diva problem
Answers at the end. On Fri, 2003-12-19 at 14:37, Daniel ANDRE wrote: > Hi Patrick, > > I have exactly the same problem with diva BRI-2M. Did you solve it? > > Regards, > > Daniel > > > Patrick a écrit: > > >Hello, > > > >I have an issue getting the chan_capi module to load in asterisk cvs > >from today. Plain 2.4.20 kernel with melware patches for the Eicon Diva > >Server Bri card. > > > >I load the modules with: modprobe -v divas divacapi > >I load the firmware with: divactrl load -c 1 -f ETSI -vd6 > > > >Output in /var/log/messages is: > >Nov 9 19:26:26 voice kernel: Eicon DIVA - DIDD table > >(http://www.melware.net) > >Nov 9 19:26:26 voice kernel: divadidd: Rel:2.0 Rev:1.13 > >Build:102-51(local) > >Nov 9 19:26:26 voice kernel: Eicon DIVA Server driver > >(http://www.melware.net) > >Nov 9 19:26:26 voice kernel: divas: Rel:2.0 Rev:1.45 Build: > >102-52(local) > >Nov 9 19:26:26 voice kernel: divas: support for: BRI/PCI PRI/PCI > >adapters > >Nov 9 19:26:26 voice kernel: divas: Diva Server BRI-2M PCI bus: > > fn: 0048 insertion. > >Nov 9 19:26:26 voice kernel: PCI: Found IRQ 14 for device 00:09.0 > >Nov 9 19:26:26 voice kernel: PCI: Sharing IRQ 14 with 00:04.2 > >Nov 9 19:26:26 voice kernel: PCI: Sharing IRQ 14 with 00:06.0 > >Nov 9 19:26:26 voice kernel: divas: bus: fn: 0048 fix > >latency. > >Nov 9 19:26:26 voice kernel: divas: Diva Server BRI-2M PCI IRQ:14 > >SerNo:1884 > >Nov 9 19:26:26 voice kernel: divas: started with major 254 > >Nov 9 19:26:26 voice kernel: divas: thread started with pid 18151 > > > >chan_capi-0.3.0 is properly installed. When I start asterisk I get in > >/var/log/messages: Nov 9 19:27:21 voice modprobe: modprobe: Can't > >locate module char-major-68 > > > >And the asterisk console says: > >[chan_capi.so] => (Common ISDN API for Asterisk) > > == Parsing '/etc/asterisk/capi.conf': Found > >NOTICE[16384]: File chan_capi.c, Line 2646 (load_module): CAPI not > >installed! > >WARNING[16384]: File loader.c, Line 305 (ast_load_resource): > >chan_capi.so: load_module failed, returning -1 > >WARNING[16384]: File chan_capi.c, Line 2733 (unload_module): Unable to > >unregister from CAPI! > >WARNING[16384]: File loader.c, Line 351 (load_modules): Loading module > >chan_capi.so failed! > > > >I have /dev/capi20 and /dev/capi20.00-19 devices with major 68. But I > >also have /dev/Divas with major 254 > >crw---1 root root 68, 0 Jan 30 2003 capi20 > >crw-r--r--1 root root 254, 0 Nov 9 19:26 Divas > > > > > >Any ideas? > > > >TIA, > >Patrick > > Hi Daniel, Make sure your card is not fighting over shared IRQs with one or more other cards. Move it around in available PCI slots to see if you can get the card its own IRQ. Turn off hardware you do not use (Parallel port, usb, serial ports, on board sound, midi/game controllers) to free up IRQs. The get yourself a plain vanilla 2.4.20 kernel from a kernel.org mirror. Here are the kernel build settings for ISDN: << ISDN subsystem >> ISDN supportM Support synchronous PPP Y Use VJ-compression with synchronous PPP Y Support generic MP (RFC 1717) Y Support BSD compression Y Support audio via ISDN Y Support AT-Fax Class 1 and 2 commands Y << Active ISDN cards >> CAPI2.0 support M Verbose reason code reporting (kernelsize +=7k) Y CAPI2.0 Middleware support (EXPERIMENTAL) Y CAPI2.0 /dev/capi support M CAPI2.0 filesystem support Y Eicon DIVA active card support M DIVA Server BRI/PCI support Y DIVA i4l supportM DIVA Capi2.0 interface support M Diva User-IDI interface support M DIVA Maint driver support M (all other options were "N") Build the kernel, install, then load the following modules: /sbin/modprobe -v kernelcapi /sbin/modprobe -v capi /sbin/modprobe -v capifs /sbin/modprobe -v divadidd /sbin/modprobe -v divas /sbin/modprobe -v divacapi /sbin/modprobe -v diva_idi Check the output in /var/log/messages to see if everything loaded ok. I have Basic Rate EuroISDN and load the firmware on the card with the following command (get the firmware from ftp.isdn4linux.org iirc) /usr/sbin/divactrl load -c 1 -f ETSI -vd6 Check the output in /var/log/messages to see if this went ok. You can monitor the activity on the D channel with the following command (get divactrl from mmm.metlware.de, needs diva_idi module loaded!) divactrl dchannel -c 1 -xlog -dmonitor Armin Schindler (author of the patches) told me that kernel 2.6 has native capi support for the Eicon cards. Very nice. When zaptel and * build on 2.6 I am definitely going to try this. Good luck, Patrick ___ Asterisk-Users mailing
RE: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
I think this would do it: Dlink DVG-1120M/H/S No experience with it -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Dawid Mielnik Verzonden: vrijdag 19 december 2003 12:02 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem) Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for the telephone. Something that would combine the functionality of a (adsl modem+) router and a SIP telephone adaptor in one box. I would appreciate any info that you might have on this. regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modprobe -r ztd-eth locks up machine...
>Did you ifdown the dynamic interfaces first ? > >Martin Yes, this still results in a crash on the box with the tor2. Is there any 'controlled' way to bring down a dynamic span? ... just for fun here is zttool (spans 3 & 4 are unused)... Alarms Span OK Tormenta 2 (PCI) Quad T1 Card 0 Span 1 OK Tormenta 2 (PCI) Quad T1 Card 0 Span 2 RED Tormenta 2 (PCI) Quad T1 Card 0 Span 3 RED Tormenta 2 (PCI) Quad T1 Card 0 Span 4 OK Dynamic 'eth' span at 'eth0/00:0A:5E:05 ... and from zaptel.conf... span=1,2,0,esf,b8zs span=2,1,0,esf,b8zs dynamic=eth,eth0/00:0A:5E:05:7E:89,24,0 > On Wed, 2003-12-17 at 10:36, john wrote: > > Hi, > > > > I have just begun working with TDMoE running between 2 fiber nics the > > dynamic span works great. In my main asterisk box's startup file I just > > 'modprobe tor2', then start asterisk. The zaptel, ztdynamic & ztd-eth > > modules all load by themselves when tor2 is loaded. If I stop asterisk then > > 'modprobe -r tor2' the tor2 module is removed but the other three remain. > > If I then 'modprobe -r ztd-eth' it causes a complete lock up on the machine. > > The remote machine does not have any zap hardware in it yet and doesn't have > > these difficulties. > > > > I know I can just restart the machine but it is in a production environment > > (soon to increase from a few to ~30 simultaneous calls) and it is nice to be > > able to make changes and > > cvs update installs without restarting. > > > > Has anyone experienced this or am I just missing a step or going in the > > wrong order? > > Unloading of modules was of such a concern that it almost didn't make it > into newer kernels. So you should probably not unload them. A production > machine should have specified service windows available. Also decent > hardware should be able to reboot fairly fast. The machine I have as our > local asterisk machine can go from reset button to accepting new calls > in under 50 seconds. Our remote machine is around 90 secs. Depending on > y our call volume, and system setup, you should be able to handle this. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFAX application
Hi all,
I have tested RxFAX application through X100P card. When Fax
arrive i obtain the next trace:
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing SetMusicOnHold("Zap/1-1", "random") in new stack
-- Executing WaitMusicOnHold("Zap/1-1", "5") in new stack
-- Started music on hold, class 'random', on Zap/1-1
-- Redirecting Zap/1-1 to fax extension
-- Stopped music on hold on Zap/1-1
== Spawn extension (default, fax, 0) exited non-zero on 'Zap/1-1'
-- Executing RxFAX("Zap/1-1", "/var/spool/asterisk/fax/uno.tif") in
new stack Changed from phase 0 to 1 Start receiving document Changed
from phase 1 to 4 Sending ident
>>> CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Store and forward Internet fax: no
Real-time Internet fax: no
Preferred octets: 256
Can receive fax
Data signalling rate: V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK Inch-based resolution preferred: no Metric-based
resolution preferred: no Minimum scan line time for higher resolutions:
T15.4 = T7.7
>>> DIS: 80 00 c6 f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
<<< TSI: 43 32 32 39 30 31 30 31 35 39 20 20 20 20 20 20 20 20 20 20 20
TSI without final frame tag Remote fax gave TSI as: "951010922" <<< DCS:
83 00 c6 f0 80 80 00 DCS with final frame tag In state 9
DCS:
Store and forward Internet fax: no
Real-time Internet fax: no
Can receive fax
Data signalling rate: V.29, 9600bps
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Minimum scan line time: 0ms
Inch-based resolution preferred: no
Metric-based resolution preferred: no
Minimum scan line time for higher resolutions: T15.4 = T7.7
Get at V.29
Changed from phase 3 to 5
Fast carrier up
Fast carrier down
Changed from phase 5 to 4
0 bad bits in trainability test
Start rx document - compression 2
Start rx page
>>> CFR: 84
HDLC underflow in state 5
Post trainability
Changed from phase 4 to 5
Fast carrier up
Fast carrier down
Fast carrier up
Fast carrier down
Fast carrier up
Fast carrier down
Fast carrier up
Equalizer state:
-7 (0.0, 0.0) -> 0.0
-6 (0.0, 0.0) -> 0.0
-5 (0.0, 0.0) -> 0.0
-4 (0.0, 0.0) -> 0.0
-3 (0.0, 0.0) -> 0.0
-2 (0.25105, 0.74039) -> 0.61121
-1 ( -0.87268,-0.36304) -> 0.89337
0 ( -1.79414,-2.01854) -> 7.29345
1 ( -0.87268,-0.36304) -> 0.89337
2 (0.25105, 0.74039) -> 0.61121
3 (0.0, 0.0) -> 0.0
4 (0.0, 0.0) -> 0.0
5 (0.0, 0.0) -> 0.0
6 (0.0, 0.0) -> 0.0
7 (0.0, 0.0) -> 0.0
Equalizer state:
-7 (0.00649,-0.03380) -> 0.00118
-6 (0.02073,-0.00508) -> 0.00046
-5 (0.03125,-0.04397) -> 0.00291
-4 (0.02094,-0.05189) -> 0.00313
-3 (0.00954,-0.02848) -> 0.00090
-2 (0.26182, 0.72429) -> 0.59315
-1 ( -0.84074,-0.41198) -> 0.87657
0 ( -1.76021,-2.05082) -> 7.30420
1 ( -0.85500,-0.38151) -> 0.87658
2 (0.22413, 0.68563) -> 0.52033
3 ( -0.03512,-0.10808) -> 0.01291
4 ( -0.02204,-0.03244) -> 0.00154
5 (0.04513, 0.11010) -> 0.01416
6 (0.05265, 0.07431) -> 0.00829
7 ( -0.01433,-0.11280) -> 0.01293
Equalizer state:
-7 (0.14017, 0.08799) -> 0.02739
-6 ( -0.18079,-0.02633) -> 0.03338
-5 (0.00565, 0.03149) -> 0.00102
-4 (0.15813,-0.07292) -> 0.03032
-3 ( -0.28991,-0.34523) -> 0.20324
-2 (0.10369, 0.59990) -> 0.37063
-1 (0.03284,-0.02606) -> 0.00176
0 ( -0.11463,-0.90551) -> 0.83309
1 (0.03712, 0.71173) -> 0.50794
2 ( -0.46280, 0.55488) -> 0.52208
3 ( -1.38062,-1.69745) -> 4.78743
4 ( -0.95295,-1.72961) -> 3.89964
5 (0.03678, 0.07704) -> 0.00729
6 (0.29737, 0.77668) -> 0.69166
7 ( -0.14636,-0.40090) -> 0.18214
Fast carrier training failed
Equali
[Asterisk-Users] Have HandyTone instock, ready to ship?
Would someone who has the HandyTone ready to ship reply offlist with price with 2day and 3day shipping to 28379? - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P errors with PRI D-channel problem
Hi All,
I need help configuring my E100P digium card. I am testing it with a marconi digital transmission analyzer. Although everything configures fine. All channels get configured and the output of cat /proc/zaptel/1 verifies that.
problems;
1. i dont know how to use the zttool for debugging purpose or any other available such tool
2. The D-channel of the PRI is shown down by asterisk whenever i try to make a call. the outpur follows:
*CLI> dial [EMAIL PROTECTED]
-- Executing Playback("OSS/dsp", "beep") in new stack
<< Console call has been answered >>
-- Playing 'beep' (language 'en')
-- Executing Dial("OSS/dsp", "zap/g2/011") in new stack
-- Called g2/011
== D-Channel on span 1 down
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
== D-Channel on span 1 down
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
WARNING[1265627968]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice
-- Hungup 'Zap/17-1'
== No one is available to answer at this time
-- Executing Congestion("OSS/dsp", "") in new stack
and the call is disconnected. but this is not always the case.when i tried it again it showed ans output like this.;
*CLI> pri debug span 1
Enabled debugging on span 1
*CLI> dial [EMAIL PROTECTED]
-- Executing Playback("OSS/dsp", "beep") in new stack
<< Console call has been answered >>
-- Playing 'beep' (language 'en')
-- Executing Dial("OSS/dsp", "zap/g2/1123") in new stack
NOTICE[1265627968]: File app_dial.c, Line 516 (dial_exec): Unable to create channel of type 'zap' == Everyone is busy at this time
-- Executing Congestion("OSS/dsp", "") in new stack
WARNING[1192437440]: File chan_zap.c, Line 5581 (zt_pri_error): PRI: Read on 49 failed: Unknown error 500
PRI got event: 8
WARNING[1192437440]: File chan_zap.c, Line 5581 (zt_pri_error): PRI: Read on 49 failed: Unknown error 500
PRI got event: 6
getting same behaviour when calling from SIP phonesI am getting suspicious behaviour by the "pri debug span 1" commandsometimes it shows the complete...but sometimes it just doesn't tell anything
one thing is for sure...the D-channel is not working.because i cant dial asterisk from the analyzer or the other way... the line has simply no errorsevrything perfect.
# Zaptel Configuration File#
span=1,2,0,ccs,hdb3
bchan=1-15dchan=16bchan=17-31
loadzone = usdefaultzone=us
; Zapata telephony interface;; Configuration file;[channels];language=en
context=itedc
immediate=yes
musiconhold=default
echocancel=no
relaxdtmf=nojitterbuffers=8switchtype=euroisdnpridialplan=unknownsignalling=pri_cpegroup=1channel=>1,2-15group=2channel=>17-21,22-31
I need help on this.anybody having similar problem..or the manufacturersregards,
M.A Ali Janjua
MSN 8 helps ELIMINATE E-MAIL VIRUSES. Get 2 months FREE*.
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[Asterisk-Users] Re: Land line vs. VoIP provider.
Robert Mann wrote: Residential Long Distance. One of my biggest pushes towards a VoIP provider was cheap long distance. Now in the U.S. at least with SBC they now have a plan for Unlimited Long Distance. The price is 30.00 a month if you do not have a couple of required features on the line already like Caller ID and a feature like 3 Way or Call Waiting etc. in which case that lowers the price to 20.00 a month. Then to top that off I have two lines in my house and by allowing them to consolidate my phone lines in to one bill they were able to offer me the Unlimited Long Distance on both lines for a single 20.00 a month charge. Now this is in the U.S. and with SBC that I know of but other companies will follow shortly if they have not already. So my question now becomes is there any reason to use a VoIP provider for outgoing calls other then cheap long distance? The other thing I was liking with some of the VoIP providers was the ability to have a incoming line from any major area so even if I live in CA I could have a NY number which is cool if you have people that call you from there or want some sort of phone spoofing (Not me of course :) ). Other then these reasons can anyone see why you would want to use a VoIP provider? I may have just saved my self some time and money choosing to use SBC now for my long distance at this one low rate. 1) International calls are still a lot cheaper. 2) Price Competition. The threat of VoIp is a major factor in inducing land carriers to keep costs reasonable. 3) Not really a good reason, but I would still cite quality. My experience with some of the cut-rate land carriers is that the quality ranges from acceptable to terrible, depending upon where the call originates. While the same is true of some VoIP carriers, the continued migration to VoIP should drive improvements, and, it is really much easier to try a number of VoIP carriers to find the one with the best quality. 4) At least for the present, it is possible AND trivial to use multiple VoIP carriers to provide failover, at little or no additional cost. This is not, as far as I know, possible with land carriers, at least not for little or no cost. 5) A single VoIP connection supports multiple simultaneous outgoing calls from any number of sites for no extra charges. No land line company can even come close to this for the price. 6) When, and if, the quality/reliability improves sufficiently, a DID line in the area code of your choice, which provides 6 simultaneous call presentations for $7.99/month, will beat any land line hands down. Summary: if you're the only caller, calling only to the US, then you might be crazy to not use a land line, especially given the deals currently available and the 911 issue (but see http://www.vonage.com/features_911.php). Even then, if you already have broadband in house (or at home), VoIP amy be an attractive alternative, if only for the control it gives you over your phone service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Land line vs. VoIP provider.
Not all VoIP providers will have Vonage's 911 issues. It's perfectly possible for a VoIP provider to provide outbound caller information to the PSAPs if they spend the time and money to do so. Stephen > Summary: if you're the only caller, calling only to the US, then you > might be crazy to not use a land line, especially given the deals > currently available and the 911 issue (but see > http://www.vonage.com/features_911.php). Even then, if you already have > broadband in house (or at home), VoIP amy be an attractive alternative, > if only for the control it gives you over your phone service. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 - can't traverse NAT?
That was it! I have spent the last week dealing with Cisco tech support on this. They even sent me a new phone. They escalated this to senior support. They couldn't get it, I couldn't get it, but you could. Thanks ever so much! Merry Christmas, Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Thursday, December 18, 2003 4:37 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Cisco 7960 - can't traverse NAT? Might be a stupid question, but is there a default gateway set on the 7960? -Original Message- From: Paul Mahler [mailto:[EMAIL PROTECTED] Sent: Thursday, December 18, 2003 7:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping anything outside the subnet the 7960 is on, that is anything on the WAN side of the router. Other devices on the subnet get out to the WAN just fine. Has anyone else had this problem or know how to fix it? Thanks in Advance, Paul Paul Mahler mail:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
Dawid Mielnik wrote: Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for the telephone. Something that would combine the functionality of a (adsl modem+) router and a SIP telephone adaptor in one box. I would appreciate any info that you might have on this. regards, Dave Take a peek at Intertex IX66+PF or look for anyone coming out with a TI AR7 based solution. It's like a $25 single chip solution. I would expect to see boxes in the $100 - $200 price range soon. If you find anything else, please let me know. I am starting to play in mid to large cat 3 environments and doing the BLEC thing bolted up to *. Very swt set up. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sphinx
--- Mauricio NuÃez <[EMAIL PROTECTED]> wrote:
>/usr/local/bin/decoder2
>#!/bin/sh
>BASE=6403
I changed this to BASE=turtle
>$S2BATCH -nbest 1 -nbestdir /tmp -verbose 0 -adcin TRUE -adcext wav ->ctlfn
>${CTLFILE} -ctloffset 0 -ctlcount 1 -datadir ${TASK} -samp >8000
>-agcmax TRUE -langwt 6.5 -fwdflatlw 8.5 -rescorelw 9.5 -ugwt 0.5 ->fillpen
>1e-10 -silpen 0.005 -inspen 0.65 -top 1 -topsenfrm 3 -topsenthresh ->7
>-beam 2e-06 -npbeam 2e-06 -lpbeam 2e-05 -lponlybeam 0.0005 -nwbeam >0.0005
>-fwdflat FALSE -fwdflatbeam 1e-08 -fwdflatnwbeam 0.0003 -bestpath TRUE
>-kbdumpdir ${TASK} -lmfn ${TASK}/${BASE}.lm -dictfn ${TASK}/${BASE}.dic
>-noisedict ${HMM}/noisedict -phnfn ${HMM}/phone -mapfn ${HMM}/map ->hmmdir
>${HMM} -hmmdirlist ${HMM} -8bsen TRUE -sendumpfn ${HMM}/sendump -cbdir >${HMM}
I put all of this on one line
>My php-test.agi:
>festival("ingrese su mensaje");
Festival was saying the text, but giving me an RTP read error. I commented it out and
now it does not.
>record("mensaje","#");
I changed this to /tmp/mensaje. Not sure where it would have recorded it otherwise.
I did a find on mensaje.wav but it did not find it anywhere until I changed it to /tmp
>$value=file("/tmp/mensaje.hyp");
I'm not seeing this file. It makes /tmp/goforward.hyp with the folowing:
debian:/tmp# more goforward.hyp
YOU WHAT TWO ARE YOU ARE
(I never said anything)
>$value=$value[0];
>$numero=trim(substr($value,strpos($value,">")+1));
>if($numero!="")
> festival("usted a seleccionado el numero $numero");
>exec($comando);
What I'm not sure about is why you are checking the value before running sphinx. This
would be getting a previous value correct? Also, I'm not sure how sphinx is getting
the file /tmp/mensaje.wav. This is the big thing that I'm missing.
The script runs for me, festival says the text (when I had it uncommented), gave a
beep, recorded the file, then quit. There was the .wav file and goforward.hyp in /tmp.
Once I understand how sphinx gets the output, I think I will have it figured out if
you could help me with that. I don't understand how sphinx knows the output file
(from the .wav) to decode it.
I appreciate your input and help very much. This is exactly what I was looking for.
Thanks,
Kevin
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Re: [Asterisk-Users] 3-way calling bug
Sorry for the lack of info...here goes - We are using CVS from earlier this week. Our phones are fxs signalling and it is connected to asterisk via a channelbank and a TE410P card. I have put a bug into the bugtracker, it's ID is 687 thanks, Derek On Thu, 2003-12-18 at 19:19, John Todd wrote: > >Hi, > > > >I discovered a problem in asterisk with the following scenerio: > > > >1) I make an outbound call > >2) Called person answers phone > >3) I hit the "flashhook" to initiate a 3-way call > >4) I hear dial tone and called person is on hold > >5) I hang up my phone > >6) called person hangs up their phone > >7) my phone starts ringing > >8) I answer and no one is there, I hang up > >9) endless loop between step 7 & 8 happens > > > >after this happens and this endless ringing loop begins asterisk cannot > >be stopped from within the console but must be killed with kill -9. > > > >Any help or insight into the matter would be greatly appreciated. > > > >Thanks, > >Derek > > What kind of phones? SIP? IAX? fxs? > > What version of CVS? What equipment? Please document a bit more, > and perhaps this might need a report in the bugtracker > (http://bugs.digium.com/) > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GotoIfTime help
Hey All,
I need to forward an extension to an other depending on the current
time but I could not get it done with GotoIfTime.
What I'm trying to do is ring on the extension 1 if time is between
8:00AM and 2:00PM and on extension 2 if is between
2:01PM 11:00PM.
exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333)
exten => 111,2,Dial(${Person1})
exten => 111,3,Dial(Hangup)
exten => 333,1,Dial(${Person2})
exten => 333,2,Dial(Hangup)
When I ring on the extension 111, the call is not being forward to the
extension 333..
And the extensions are all in the same context.
regards
Oz
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RE: [Asterisk-Users] Re: Land line vs. VoIP provider.
What about having your VoIP gateway system placing a 911 call to the 911 answering center in the appropriate region and when the 911 operator answers, have a message say "This is a 911 call from 123 Main Street, Nowhere Nebraska" then connect the caller to the 911 operator. Legal? Maybe. Dunno. Just a random thought that I came up with on the way to the aforementioned middle of nowhere, Nebraska. > > Not all VoIP providers will have Vonage's 911 issues. It's perfectly > possible for a VoIP provider to provide outbound caller information to > the PSAPs if they spend the time and money to do so. > > Stephen > > >> Summary: if you're the only caller, calling only to the US, then you >> might be crazy to not use a land line, especially given the deals >> currently available and the 911 issue (but see >> http://www.vonage.com/features_911.php). Even then, if you already > have >> broadband in house (or at home), VoIP amy be an attractive > alternative, >> if only for the control it gives you over your phone service. >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sphinx
Hi,
If you put BASE=turtle, then the script search for
/var/lib/asterisk/model/lm/turtle/turtle.ctl, and the content of this file is
a list of files to process, without the extension, because this is provided
via the parameter -adcext of sphinx2-continuous.
the turtle dir is the demo provided for sphinx, and my script is practically a
copy paste, modifyied to explore distincts ${BASE} dir.
Inside ${BASE}, i had a control file ${BASE}.ctl, with the line:
/var/lib/asterisk/sounds/mensaje
hardcoded.
yo get goforward.hyp because turtle.cl constain goforward.16k as input file.
To work with asterisk, i put the -sample option at 8000 , because that is the
wav sampling. The turtle example work at 16000, and that's explain because
you get a strange result.
I'm executing wrongly the command ? I' will check that. I'm trying to get
asterisk work with sphinx from yesterday :-), but almost is working (slow,
but working)
Saludos
Mauricio NuÃez
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Re: [Asterisk-Users] GotoIfTime help
about GotoIfTime you have:
>show application GotoIfTime
-= Info about application 'GotoIfTime' =-
[Synopsis]:
Conditional goto on current time
[Description]:
GotoIfTime(|||?[[context|]extension|]pri):
If the current time matches the specified time, then branch to the
specified extension. Each of the elements may be specified either as
'*' (for always) or as a range. See the include syntax.
and it have to be something like that:
[default]
exten => 111,1,GotoIfTime(08:00:00-14:00:00,*,*,*?extension_1,1,1)
exten => 111,2,GotoIfTime(14:00:01-23:00:00,*,*,*?extension_1,101,1)
exten => 111,3,Hangup
[extension_1]
exten => 1,1,Dial(${Person1})
exten => 1,2,Hangup
exten => 101,1,Dial(${Person2})
exten => 101,2,Hangup
Lubo
Osvaldo Mundim wrote:
Hey All,
I need to forward an extension to an other depending on the current time
but I could not get it done with GotoIfTime.
What I'm trying to do is ring on the extension 1 if time is between
8:00AM and 2:00PM and on extension 2 if is between
2:01PM 11:00PM.
exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333)
exten => 111,2,Dial(${Person1})
exten => 111,3,Dial(Hangup)
exten => 333,1,Dial(${Person2})
exten => 333,2,Dial(Hangup)
When I ring on the extension 111, the call is not being forward to the
extension 333..
And the extensions are all in the same context.
regards
Oz
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Re: [Asterisk-Users] GotoIfTime help
Hi!
gotoif usually takes a priority as label, not an extension! See below.
Never tried if you can also use the notation "context,extension,priority"
instead of just "priority", but it might work. Just try it.
> I need to forward an extension to an other depending on the current
> time but I could not get it done with GotoIfTime.
>
> What I'm trying to do is ring on the extension 1 if time is between
> 8:00AM and 2:00PM and on extension 2 if is between
> 2:01PM 11:00PM.
>
> exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333)
> exten => 111,2,Dial(${Person1})
> exten => 111,3,Dial(Hangup)
>
> exten => 333,1,Dial(${Person2})
> exten => 333,2,Dial(Hangup)
exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?4:2)
exten => 111,2,Dial(${Person1})
exten => 111,3,Dial(Hangup)
exten => 111,4,Goto(default,333,1)
exten => 333,1,Dial(${Person2})
exten => 333,2,Dial(Hangup)
Note: If you use goto() or gotoif() with just one label then you'll see a
warning in /var/log/asterisk/messages about the 2nd label missing. That's
why I prefer to always specify both labels.
Cheers, Philipp
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[Asterisk-Users] PRIMARY=wcfxo?
I was wondering is anyone can tell more about the following lines in the zaptel makefile: #PRIMARY=wcfxsusb PRIMARY=torisa #PRIMARY=wcfxo I am using 2 zaptel X100P's, and wondering if I should change this in my Makefile to: #PRIMARY=torisa PRIMARY=wcfxo Just a thought. any info is appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIfTime help
All right...
Its working now.
thank you very much!
regards
Oz
On Dec 19, 2003, at 3:02 PM, Philipp von Klitzing wrote:
Hi!
gotoif usually takes a priority as label, not an extension! See below.
Never tried if you can also use the notation
"context,extension,priority"
instead of just "priority", but it might work. Just try it.
I need to forward an extension to an other depending on the current
time but I could not get it done with GotoIfTime.
What I'm trying to do is ring on the extension 1 if time is between
8:00AM and 2:00PM and on extension 2 if is between
2:01PM 11:00PM.
exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333)
exten => 111,2,Dial(${Person1})
exten => 111,3,Dial(Hangup)
exten => 333,1,Dial(${Person2})
exten => 333,2,Dial(Hangup)
exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?4:2)
exten => 111,2,Dial(${Person1})
exten => 111,3,Dial(Hangup)
exten => 111,4,Goto(default,333,1)
exten => 333,1,Dial(${Person2})
exten => 333,2,Dial(Hangup)
Note: If you use goto() or gotoif() with just one label then you'll
see a
warning in /var/log/asterisk/messages about the 2nd label missing.
That's
why I prefer to always specify both labels.
Cheers, Philipp
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Re: [Asterisk-Users] Sphinx
Hi,
--- Mauricio NuÃez <[EMAIL PROTECTED]> wrote:
>Inside ${BASE}, i had a control file ${BASE}.ctl, with the line:
>
>/var/lib/asterisk/sounds/mensaje
>
>hardcoded.
Ok, got that.
>To work with asterisk, i put the -sample option at 8000 , because that >is the wav
>sampling. The turtle example work at 16000, and that's >explain because you get a
>strange result.
Got that.
>I'm executing wrongly the command ? I' will check that. I'm trying to >get asterisk
>work with sphinx from yesterday :-), but almost is >working (slow, but working)
It complains that the wav file does not exist the first time it runs. I had to
quickly patch things together to restart it. Getting ready to head out to school and
get groceries. I will look further into that part. Have to get a microphone to test
it further. It outputs A now which should be good.
The next big problem I see is that every time you run the script, it uses the same
file. This will need to be modified to be a unique filename for each
caller/iteration. You are doing very well for your second day with Sphinx. A lot
better than me. Hopefully between the two of us, we can get that problem fixed. It's
a step in the right direction.
Thanks,
Kevin
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Re: [Asterisk-Users] Re: * with RADIUS
On Thu, 11 Dec 2003, Jeremy McNamara wrote: > Chandra wrote: > > Explain why you think you really need RADIUS Accounting? Why not talk > right to the database itself and save yourself that unneeded > complication and points of failure. Jerey, ISP's integrating Asterisk could utilize their existing radius billing packages to easily bill customers for their usage. I, personally, would love this feature, and Radius is widely deployed and stable. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911 settings.
I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine. But when they call 911 they get our main address and not the other address's. So we need to be able to give the correct address to the 911 call! This is just for our locations and not for reselling our Asterisk server! - \ \\_ Ariel Batista //IS Director / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: * with RADIUS
On Thu, 11 Dec 2003 [EMAIL PROTECTED] wrote: > > Explain why you think you really need RADIUS Accounting? > > Why not talk > > right to the database itself and save yourself that unneeded > > complication and points of failure. > > > > I know this has come up before, and in a perfect world, where * was the > primary app, you don't need RADIUS. In enterprise environments where > RADIUS accounting is already embedded into other aspects of the > workflow, it would be beneficial. > > Understand* boxes are in real live actual production now. Once you > leave the vacuum of the lab, there are going to be things like this that > come up. And many will be for good reasons. Others will be for crappy, > legacy reasons. Both scenarios are valid in the real world. I wrote a small application that measures traffic totals for Websites, Switch ports virtually anything else that can be accessed via SNMP. That application then takes the totals and sends them as a Radius Accounting packets to our Radius Servers, which store the data in a SQL backend. Our billing system (Platypus) can then generate bills for customers based on usage.. in this case Gigabytes of Data transferred. For our dial-up customers is based on a certain block of hours with minutes charged at a specific rate over the limit. It is a system that makes it trivial to assign a value to some data point and create an itemized invoice for it. I am in the process of deploying an Asterisk server to provide voice services to about 20 phones. It would be nice to be able to use our existing billing system to simply charge people for the minutes they have used and/or use the block pricing models that we already have in place (I.E. First 1000 minutes free, .03 each additional) without having to use another billing system and import/export data. For me, having CDR data delivered as a Radius accounting packet would save me tons of development time, and many hours of additional work implementing a paralell billing system for the express purpose of accurate billing. So.. I'll be happy to help out in the effort, but my C coding skills are rustier than hell, and more than likely I'll break many things along the way. ;) On the other hand, if someone has a simple, open-source, SQL based billing, invoicing, CDR management program, I am all ears! If I can spend an hour or two setting up something that works well, with little effort, it would buy me lots of time. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * with RADIUS
On Thu, 11 Dec 2003, Andrew Thompson wrote: > > Understand* boxes are in real live actual production now. Once you > > leave the vacuum of the lab, there are going to be things like this that > > come up. And many will be for good reasons. Others will be for > > crappy, legacy reasons. Both scenarios are valid in the real world. > > > > Can someone give me an idea exactly what things are intended to be tested > via RADIUS, or some other AAA system? > > Are we talking about building SIP/IAX/H323 entries from RADIUS? > > At this point, I'm not really worried about call detail, as that's something > that * already can dump to a database, it can be adapted to dump back to any > service. (Unless this is really the primary objective of the whole RADIUS > discussion.) For me, it is. Being able to take CDR data (Called Number, Port, Start Time, Stop Time, Extension etc...) and dump them to a Radius server means I don't have to implement a seperate billing program to charge for usage! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * with RADIUS
On Thu, 11 Dec 2003, Andrew Thompson wrote: > - Original Message - > From: "Doug Shubert" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, December 11, 2003 11:48 AM > Subject: Re: [Asterisk-Users] Re: * with RADIUS > > > > Hello, > > We use RADIUS with a MySQL backend database server for dialup > > authentication. > > Because our accounting system is XML based, I would prefer to use one AAA > > (i.e RADIUS) > > server to provision and validate our VoIP UA's. LDAP is another AAA > solution > > we are looking at using. > > Of course a direct SQL connection from * would still work with the > backend. > > Doug > > So, you would like to build at least the extension list portion of a > Dialplan from RADIUS on boot and as it changes, correct? > > You would expect SIP/H323/IAX/zap-destinations to be created(or activated) > from RADIUS as well, right? That is interesting. I've been approaching it from the Radius ACCOUNTING perspective, not neccessarily the Authentication perspective, but it does open up some interesting possibilities that I haven't fully wrapped my brain around yet. Let me think on it, drink a few beers, play with my * server and get back to you. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Land line vs. VoIP provider.
Hi Stephen, Interesting >6) When, and if, the quality/reliability improves sufficiently, a DID >line in the area code of your choice, which provides 6 simultaneous call >presentations for $7.99/month, will beat any land line hands down. I did not know that one connection can have many simultaneous call presentations. I just tried with my iconnect account, So while I was calling in to an * extension, I tried calling the same iconnect number again. It didn't give me an engage tone, infact it came to the * as well. Six simultaneous presentations you mentioned here is for VoicePulse, right ? Do you know the limitation for Iconnect ? I can't use VoicePulse because I live in California, and need California numbers. Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Land line vs. VoIP provider.
James Sharp wrote: What about having your VoIP gateway system placing a 911 call to the 911 answering center in the appropriate region and when the 911 operator answers, have a message say "This is a 911 call from 123 Main Street, Nowhere Nebraska" then connect the caller to the 911 operator. Legal? Maybe. Dunno. Just a random thought that I came up with on the way to the aforementioned middle of nowhere, Nebraska. As far as I know, that's more or less what Vonage does do. You register your address with them, and if a 911 call comes in, they forward the location info to the 911 service in your area. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 settings.
- Original Message - From: "Ariel Batista" <[EMAIL PROTECTED]> To: "Asterisk User List" <[EMAIL PROTECTED]> Sent: Friday, December 19, 2003 4:06 PM Subject: [Asterisk-Users] 911 settings. > I would like to know if anyone has come up with a script for 911 dialing > rules that put correct information on our locations. We have our office > in 3 different building one being our production & shipping dock. It is > almost 2 blocks away. We are connected with Ethernet Wireless between > the buildings and have Sip phones setup in the other 2 locations. All > the phones are working just fine. But when they call 911 they get our > main address and not the other address's. So we need to be able to give > the correct address to the 911 call! This is just for our locations and > not for reselling our Asterisk server! The 911 office is most likely retreiving the address off of the line that is placing the call. Do you have any voice lines in the other buildings? I would consider a line siege device and FXO attached to a fax or security system line in the other buildings. Route the dialed 911's out over the local pots line and they will get the correct address. I don't know if you can attach an address any other way. You could try sending a different callerid, but if they are all billed as being in the main building, that's probably the address they'll get. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Land line vs. VoIP provider.
SW wrote: Hi Stephen, Interesting 6) When, and if, the quality/reliability improves sufficiently, a DID line in the area code of your choice, which provides 6 simultaneous call presentations for $7.99/month, will beat any land line hands down. I did not know that one connection can have many simultaneous call presentations. I just tried with my iconnect account, So while I was calling in to an * extension, I tried calling the same iconnect number again. It didn't give me an engage tone, infact it came to the * as well. Six simultaneous presentations you mentioned here is for VoicePulse, right ? Do you know the limitation for Iconnect ? I can't use VoicePulse because I live in California, and need California numbers. Thanks SW Sorry, I should have been more explicit. Yes, I meant with VoicePulse and as far as I know, with Nufone too. These are the only ones I have checked explicitly (the IAX providers). Also, the number of simultaneous presentations may bary from provider to provider. Vonage may be able to do the same thing as well, but you would have to verify that with them, or someone on the list may also know. As far as I know, iconnect explicitly disallows multiple call presentations. The iconnect thing was discussed on the list a month or two back. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip registration change!
I have a question on SIP devices that are setup and working but you change the login name and contents to them why does asterisk need to be shut down and restarted for them to work? I have reloaded extensions and done a reload command. But the updated sip phones do not work until I shut down and restart asterisk. Is there any other way to update them without restarting the system? Since the system is used allot it's hard to find a time to restart it. - \ \\_ Ariel Batista //IS Director / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX phone busy
I've configured the DIAX phone. It registers with the * server, and I am able to make calls from DIAX. However, when I try to call the DIAX phone from another phone, I get a busy signal. My extensions.conf: exten => 70,1,Dial(IAX/mike/mike,30,tr) # IAX Mike exten => 70,2,Voicemail(u70) exten => 70,102,Voicemail(b70) and my iax.conf: [mike] type=friend username=mike host=dynamic secret=pass1 ;auth=md5,plaintext,rsa ;mailbox=70 context=local permit=0.0.0.0/0.0.0.0 Is my configuration correct? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 200 and * issues
Hello All, My SNOM 200 phone keeps generating the following message on the * console: Notice [11127005368] : File chan_sip.c Line 5394 (handle_request) : Unknown sip command 'Publish' from '192.168.1.17' What does this mean and how do I remedy the problem? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 "One day in your life shouldn't be a problem" - 54-40 from One Day ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Land line vs. VoIP provider.
> As far as I know, iconnect > explicitly disallows multiple call presentations. The iconnect thing was > discussed on the list a month or two back. I red that discussion, it was more on multiple outgoing calls. I noticed with Iconnect, the sip invite message always comes as the [EMAIL PROTECTED] So if I use two PSTN numbers to call same Iconnect provided number, it arrives at asterisk as two different calls. Iconnect seems like completely depends on the SIP response from * or any SIP end point to figure out the call status (busy or on call). That is why I think, it is IMPOSSIBLE to set the * to receive a call coming from Iconnect to a particular context, other than general section, in it's sip.conf. (I didn't hear from any one who got this working) I think this is how, fwd and others are offering access numbers in many US States. Using a single number many callers can get to their (fwd) network. This is my observation, please correct me if I am wrong ? SW > -Original Message- > From: Stephen R. Besch [mailto:[EMAIL PROTECTED] > Sent: Friday, December 19, 2003 2:29 PM > To: [EMAIL PROTECTED] > Cc: [EMAIL PROTECTED] > Subject: Re: Land line vs. VoIP provider. > > > SW wrote: > > Hi Stephen, > > > > Interesting > > > > > >>6) When, and if, the quality/reliability improves sufficiently, a DID > >>line in the area code of your choice, which provides 6 simultaneous call > >>presentations for $7.99/month, will beat any land line hands down. > > > > > > I did not know that one connection can have many simultaneous call > > presentations. > > > > I just tried with my iconnect account, So while I was calling in to an * > > extension, I tried calling the same iconnect number again. It > didn't give me > > an engage tone, infact it came to the * as well. > > > > Six simultaneous presentations you mentioned here is for > VoicePulse, right ? > > > > Do you know the limitation for Iconnect ? I can't use > VoicePulse because I > > live in California, and need California numbers. > > > > Thanks > > > > SW > > Sorry, I should have been more explicit. Yes, I meant with VoicePulse > and as far as I know, with Nufone too. These are the only ones I have > checked explicitly (the IAX providers). Also, the number of simultaneous > presentations may bary from provider to provider. Vonage may be able to > do the same thing as well, but you would have to verify that with them, > or someone on the list may also know. As far as I know, iconnect > explicitly disallows multiple call presentations. The iconnect thing was > discussed on the list a month or two back. > > Stephen R. Besch > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interconnecting Panasonic KX-TD1232 digital PBX and *
Dan wrote : > Subject: [Asterisk-Users] Interconnecting Panasonic KX-TD1232 digital > PBX and * > Hi all, > > There is someone with some experience interconnecting a Panasonic digital > PBX (KX-TD1232) with Asterisk? > Ehh, what exactly do you want to do? I've got * 'interfaced' via the ISDN S0 bus of my PBX. I'm not exactly sure if the Panasonic offers S0 ISDN, but you could of course also use an analog extension port. I can be called from the phones on my PBX and I can call internal phones from * clients (sip and iax). Arnold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE
Hi all, I am looking at setting up a TDMoE link between * boxes and am having a rough time locating and documentation or configuration examples. I have gotten far enough to get the dynamic link up between boxes, but not sure where to go from here. I'm not even sure which modules need to be loaded. I've loaded ztd-eth and ztdynamic, but saw something about ppp_generic so I'm not sure if that's another one to include or not. I've tried several different config changes on the boxes to create the actual * PRI, but have failed in miserable style! When I run ztcfg with the dynamic span configured as: dynamic=eth,eth0/00:90:27:93:C0:60,24,0 e&m=3-26 it complains: ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Any guidance on this, and the zapata.conf that finishes this, would be much appreciated! Thanks! Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Excellent service from vendor
We often hear about the problems with vendors, but not as often when a vendor does more than expected. Last night I placed an order for a Sipura ATA with http://www.voxilla.com/ Today they called me to explain that they had just run out of inventory, and would be happy to cancel/refund my order, or give me a free upgrade to next day shipping when they receive more inventory next week. This level of service is unusual these days -- at least among the vendors I've been dealing with. Jim James H. Thompson[EMAIL PROTECTED]
[Asterisk-Users] Level(3) SIP termination services?
Anyone investigated the new service offerings from Level(3) in the last few months? They claim to be using ENUM and SIP - see http://www.level3.net/2192.html for details. Any idea of their pricing model for mid-sized enterprise applications or call centers for origination/termination? More specifically, do they interoperate with Asterisk? Some providers insist on certain hardware that speaks SIP flavor-of-the-month. I could call them to find out, but I suspect that this list will have far more clue than the Level(3) sales weasel that I'd get on the phone and who would want to waste a few days of my time asking stupid questions of me. Replies off-list, if you feel it necessary. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Land line vs. VoIP provider.
At 3:42 PM -0800 12/19/03, SW wrote: > As far as I know, iconnect explicitly disallows multiple call presentations. The iconnect thing was discussed on the list a month or two back. I red that discussion, it was more on multiple outgoing calls. I noticed with Iconnect, the sip invite message always comes as the [EMAIL PROTECTED] So if I use two PSTN numbers to call same Iconnect provided number, it arrives at asterisk as two different calls. Iconnect seems like completely depends on the SIP response from * or any SIP end point to figure out the call status (busy or on call). That is why I think, it is IMPOSSIBLE to set the * to receive a call coming from Iconnect to a particular context, other than general section, in it's sip.conf. (I didn't hear from any one who got this working) I think this is how, fwd and others are offering access numbers in many US States. Using a single number many callers can get to their (fwd) network. This is my observation, please correct me if I am wrong ? SW [snip] You should be able to hand off calls from Iconnecthere to whatever context you want, based on the IP address of the host that sends you the call. The number in the call is irrelevant. To answer a few of the other threads that seem to be unraveling out of this Subject: - anyone that offers "all-you-can-eat" dialplans will never, ever let you have more than one session outbound at a time. If you find anyone that does, let me know. I have some pipelining I need to test out... - Voicepulse and Nufone allow simultaneous calls, since they charge per minute. As far as they're concerned, theoretically, you could fire up 200 sessions at once - that's great for them, they get paid on all of 'em. - It is the rumor that Vonage does not support simultaneous sessions due to back-office billing system problems which prevent their systems from discovering that there are multiple calls happening on the same number/account. This leads me to conclude that they do not have a billing system that is fully functional, thus their plan of "all you can eat" is not 100% market-driven, thus their policy of "you don't get the password to your box or you'll cram us and it will be weeks before we realize we're bleeding." Someone from Vonage who is reading this list (I'm sure there are several of you) may reply to refute this chain of assumptions if you wish, but I doubt you will. :-) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P connected to Cisco
Hi All, I wish to connect * to a Cisco using a E100P board. When I load the driver I got this error message: -bash-2.05b# modprobe wct1xxp ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows Cisco configuration: isdn switch-type primary-qsig isdn voice-call-failure 0 controller E1 2 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 interface Serial2:15 no ip address isdn switch-type primary-qsig isdn overlap-receiving T302 2000 isdn incoming-voice modem isdn T310 4 isdn send-alerting no cdp enable voice-port 2:D cptone BR I configured my /etc/zapata.conf: span=1,0,0,ccs,hdb3 nethdlc=1-15 Any clue? Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - Ringback
I am new to the sip side of things and have a question regarding ringback. I don't hear ringback when using the sjphone softphone when dialing internal extensions. It's fine when dialing outside over the pstn. Is this a issue of the softphone, configuration or sip in general? Thank you, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Zaptel Load on Startup
After searching the archives for a while, I couldn't find any easy way to get everything loaded on startup. So I decided to take a stab at writing some notes on what I've found. If everyone chips in, maybe we can make that part easier for new users! Both the Zaptel and Asterisk packages have a make option called config (make config). This option adds an entry to your init.d directory for the applicable package, and registers it with the system. If you edit the zaptel file that's created, you'll find a string called MODULES that has most of the zaptel modules in it. There's also a string called RMODULES that appears to unload all modules loaded my MODULES. If you remove a module from one, you should probably remove it from the other. Now who wants to correct, comment, or add to this?! Sean <>
Re: [Asterisk-Users] 911 settings.
At 5:26 PM -0500 12/19/03, Andrew Thompson wrote: > I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine. But when they call 911 they get our main address and not the other address's. So we need to be able to give the correct address to the 911 call! This is just for our locations and not for reselling our Asterisk server! The 911 office is most likely retreiving the address off of the line that is placing the call. Do you have any voice lines in the other buildings? I would consider a line siege device and FXO attached to a fax or security system line in the other buildings. Route the dialed 911's out over the local pots line and they will get the correct address. I don't know if you can attach an address any other way. You could try sending a different callerid, but if they are all billed as being in the main building, that's probably the address they'll get. Aye, there's the rub. I'll be brief on 911 and VoIP, but it's a topic about which I could complain several days. The problem is threefold: 1) which 911 center do we connect this VoIP user to? 2) what caller ID do we give to the 911 center? 3) how do we get street address to the 911 center for dispatch? Some providers Whose Names Will Go Unmentioned Here have come up with a system that they are selling to large VoIP service providers/IPSCP's. Without violating any NDA's (see references for public information sources) here is the general view for each option: 1) If you have the address of the customer, this E911 provider allows you to send (via XML over HTTP, apparently) the address information of the customer and their phone number to the central database. The central database then feeds back to you the ten-digit phone number that lets you into the "normal" phone line for the 911 center. It's then your job, as the IPCSP or PBX provider, to send the call the correct path to get to that ten-digit number. 2) Good question, especially with VoIP phones. No "true" solution exists for this across all providers; it depends on implementation. If you have DID's associated with each station, you're in luck. If you are using twelve digit random extensions, all homing out of one single DID for outbound caller ID, then you'll have to come up with some clever way around that, won't you? My favorite is a temporary mapping of some small pool of DID's to the last SIP URI's that made 911 calls - you have maybe a block of 1000 numbers that you round-robin and attach to 911 calls so that when the PSAP calls back the DID, they get auto-mapped to the SIP URI of the original caller. Maybe two or three days later, that number gets re-mapped somewhere else... I haven't discussed this idea with any PSAP operators, but I'd be interested in opinions from the list as to it's usefulness. 3) You send your customer list and address information via some type of update to a central repository. That repository is hooked into the brains of some of the 911 systems across the country (but potentially not all of them.) The database is called ALI, or Automatic Location Information. Updating ALI unless you are a large phone company (or ALI service provider) is very difficult, and is normally a very expensive proposition. Small or medium PBX systems like Asterisk will be left far, far behind in this because the users are typically low-budget and don't have the time to waste building "relationships" with fussy sales- or paperwork-heavy organizations which can provide those services. In any case, not always is it the case that the ALI provider can push the street address information all the way out to the PSAP, and also there is often a logical disconnect between the ALI data and the phone call itself if you're in a VoIP environment. Keeping the addresses updated in the ALI is 100% the problem of the ITCSP - there are at least four methods of this that have been discussed previously to solve this at a policy and technology level, all with various faults and favors, a combination of which I'm sure could see "acceptable" accuracy given the technology gap. Yes, it's much more complex than this, but I said I would be brief... Solution: What is needed is a VOIP-CLUEFUL provider of E911 PSAP mapping data, ALI transfer, and maybe even SIP call forwarding and processing to PSAPs. They should be low-priced on a monthly basis ($1 a line?), have IP connectivity to customers across the public Internet, be open-source for their client-side implementations, and provide (possibly) reverse call mapping for customers via DIDs. I would even support the construction of a non-profit compan
RE: [Asterisk-Users] Re: Land line vs. VoIP provider.
It's certainly not _illegal_ in any way that I can think of, and I expect that anything is better than no information at all. Sounds like a good idea. The only shortcoming I can think of is the lack of ability for the PSAP to hear the address more than once without making an unacceptable delay in connecting the call if you read it back a few times. This is yet another reason for Asterisk to support some type of in-band DTMF detection from within Dial and jumping using something other than the !$)%)$&!%^ "pound" (#) key. "This is an emergency call from 123 Main Street, Nowhere, Nebraska, zip code 12345. You will now be connected to the calling party. Press the star key at any time to hear this address again." The trick would be to have Dial jump out without hanging up, process the possible dialed match string, run the extensions, play back the message again, and jump back into Dial and re-link the legs of the call. Hmm... [emergency] exten => 911,1,Dial(Zap/g1/911,jA(read-the-address-file.wav)) exten => 911,2,Hangup exten => *,1,Playback(read-the-address-file.wav) exten => *,2,Goto(911,1) New Dial Modifiers: 'J' - Allow calling party to press a single digit to jump out of Dial. The called party hears music on hold (if selected) or silence. If 'J' is specified and there was a previous incantation of a Dial with "J", then the leg is simply reconnected to the holding caller without re-dialing the channel. If the digit specified is not in the context, no action is taken. Only one jump may be specified in a string of Dials; latest jump wins, and previously Jumped Dials are hung up on if more than one is nested. 'j' - Same as "J", but allows called party to press a digit. JT What about having your VoIP gateway system placing a 911 call to the 911 answering center in the appropriate region and when the 911 operator answers, have a message say "This is a 911 call from 123 Main Street, Nowhere Nebraska" then connect the caller to the 911 operator. Legal? Maybe. Dunno. Just a random thought that I came up with on the way to the aforementioned middle of nowhere, Nebraska. Not all VoIP providers will have Vonage's 911 issues. It's perfectly possible for a VoIP provider to provide outbound caller information to the PSAPs if they spend the time and money to do so. Stephen Summary: if you're the only caller, calling only to the US, then you might be crazy to not use a land line, especially given the deals currently available and the 911 issue (but see http://www.vonage.com/features_911.php). Even then, if you already have broadband in house (or at home), VoIP amy be an attractive alternative, >> if only for the control it gives you over your phone service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 settings.
Andrew Thompson wrote: - Original Message - From: "Ariel Batista" <[EMAIL PROTECTED]> To: "Asterisk User List" <[EMAIL PROTECTED]> Sent: Friday, December 19, 2003 4:06 PM Subject: [Asterisk-Users] 911 settings. I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine. But when they call 911 they get our main address and not the other address's. So we need to be able to give the correct address to the 911 call! This is just for our locations and not for reselling our Asterisk server! The 911 office is most likely retreiving the address off of the line that is placing the call. Do you have any voice lines in the other buildings? I would consider a line siege device and FXO attached to a fax or security system line in the other buildings. Route the dialed 911's out over the local pots line and they will get the correct address. I don't know if you can attach an address any other way. You could try sending a different callerid, but if they are all billed as being in the main building, that's probably the address they'll get. Another solution that might work is to ask the phone company to change the address that they give the PSAP on one of your phone numbers to the other building, and then use that for 911. I don't know how big of a customer you are for your phone company, but if you have more than a token number of lines they'll hopefully go for it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 settings.
On Fri, 19 Dec 2003, Nick Bachmann wrote: > I don't know how big of a customer you are for your phone company, but > if you have more than a token number of lines they'll hopefully go for it. Another option is to call the non-emergency number of the dispatch center and explain "this one number/address could actually mean someone is calling from either this location or the one down the street... Make sure you get this information from the caller.". Typically they can add some comments to their database at the dispatch center (they typically use this feature for making note of things like "site stores 3 million gallons of highly explosive substance", which the phone company doesn't keep in their databases). It's not as good as knowing exactly where the call is coming from, but it is a start. It might be good for people that have non-local phone numbers, too, and 911 is translated to the non-emergency phone number. If you call them up and talk to them, they are typically glad to help. They often are willing to help you test the actual 911 part of your dial-plan, too (are you *SURE* you haven't screwed that up? The only way to find out is to test this - WITH THE DISPATCH CENTER'S COOPERATION). (this isn't a bad thing to do even if you are using a land line with supposedly the right address - all computer databases are not 100% accurate...) -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit the Directory Application?
On Saturday, 13 December, 2003 11:16, Tilghman Lesher wrote: > ... > > Directory does not need an escape condition. If you fail to enter > anything within the allotted time (see ResponseTimeout), you jump > to the t extension. > That makes for a rather ill solution for the poor fool (like me, often) who accidently enters the directory and starts pounding all of the usual escape keys because he is impatient. Okay, so I am a little restricted by temper... > > In a production environment, it is far better to take them as a > > proof-of-concept/development base and customize them to your overall > > setup than to use them out of the box. > > We use Voicemail() out of the box in multiple production environments. Yes. Unfortunately, so have I. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to detect process 256 frames
> >WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to > >detect process 256 frames > Do not try to do inband DTMF on G.729 Can we wiki-fy this? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
