Re: [Asterisk-Users] OT?: International number parsing

There is second single digit code - 7 (Russia). On Sat, 2006-01-28 at 18:41 +0100, Francesco Peeters (Asterisk) wrote: Only one country has a single digit code: USA = 1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-User

Re: [Asterisk-Users] spandsp Error

Line 103 in Makefile has multiple spaces at the beginning instead of TAB character. On Mon, 2006-01-23 at 10:19 +0800, Ronald Wiplinger wrote: I cannot see it make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/pbx' /bin/sh: curl-config: command not found make[1]: Entering

Re: [Asterisk-Users] Alphanumeric pattern match in extensions.conf

The match doesn't work because "n" in "conf" will never match to the letter "n" (it's a pattern for a digit). try _co[n]f. instead. On Fri, 2006-01-06 at 10:33 -0800, Dan Austin wrote: > I need to match an incoming call based on a prefixed string, and this > solution was suggested to me some time

Re: [Asterisk-Users] Via Epia

See http://www.epiawiki.org/wiki/tiki-index.php?page=EpiaInstallingGentoo regarding CFLAGS settings for different VIA CPUs. On Sat, 2005-12-10 at 11:57 +, Roger Hill wrote: Aha! I was getting the same error and could not figure out why. My CPU is a VIA Samuel. So it's a VIA thing?? R

Re: [Asterisk-Users] Via Epia

I run * on VIA EPIA M1 on gentoo. Here is my ~/.asterisk.makeopts: K6OPT  = -DK6OPT DEBUG= ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk PROC=i686 On Sat, 2005-12-10 at 12:36 +0100, Maciej Kietlinski wrote: >Does anyone has some experience in installing * on Via Epia. I am >strugglin

Re: [Asterisk-Users] GotoIf always goes to true?

Shouldn't the _expression_ be GotoIf($[${T}<3]?3:7) On Fri, 2005-11-18 at 15:39 -0800, Andy Kuo wrote: Hi all,   I recently found GotoIf not working right in my extensions.conf, so I write a simple test and test it on my newly installed v1.2 box. However, in all cases,

Re: [Asterisk-Users] Asterisk 1.2 Released!

cd "top level asterisk source directory (where UPGRADE.txt is)" patch -p0 does the following patch work for 1.2? how to apply it to 1.2? ( I > am not a programmer, don't know how to use .diff file). > > http://bugs.digium.com/view.php?id=5374 > silence-suppression-2.diff > > > On 11/17/05,

Re: [Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end

On Wed, 2005-11-16 at 19:31 +1300, Matt Riddell wrote: > Sergey Okhapkin wrote: > > Already supported (simple patch exists). > > http://bugs.digium.com/view.php?id=5374 > > Which? silence-suppression-2.diff > It's already supported in Asterisk or you can patch As

Re: [Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end

Already supported (simple patch exists). http://bugs.digium.com/view.php?id=5374 On Wed, 2005-11-16 at 12:32 +1300, Matt Riddell wrote: Asterisk guy wrote: > dropping extra frame of G.729 since we already have a VAD frame at the end- Turn off VAD, it is not supported by Asterisk. _

Re: [Asterisk-Users] Comments in AEL files?

//comment AEL ignores any text from // till the line end. On Mon, 2005-11-14 at 07:39 -0800, Ed Greenberg wrote: > Any way to comment out a line (or some text) in an AEL file? > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asteris

Re: [Asterisk-Users] IAXy echo?

Lower speaker volume on the phone connected to IAXy. On Mon, 2005-11-14 at 07:21 -0600, Mike Hammett wrote: > I've got two customers on the same broadband provider. Same Asterisk > box on my end. Same CLEC. > > One has an IAXy and the other has an Asterisk box with an array of > devices (Grand

RE: [Asterisk-Users] Asterisk realtime extensions context inclusion

That's what macro is useful for. Don't include these common contexts, but convert them to macros and call these macros from user's dialplan. Macros in asterisk dialplan are close to subroutines rather than to C-style #define. On Mon, 2005-11-14 at 07:44 +, Daniel Clark wrote: Thanks f

Re: [Asterisk-Users] WARNING[3035]: Invalid priority/label ' ' at line 17

Post line 17 of your extensions.conf file. On Sat, 2005-11-12 at 18:47 -0600, Greg Blakely wrote: I just recently upgraded to the latest HEAD, and am now getting the following warning: -- Including context 'fromcnet' in context 'pots' Nov 12 18:45:17 WARNING[3035]: pbx_config.c:1697 pb

Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1

Asterisk sends OPTIONS message if the device have "qualify=NNN" option set. On Fri, 2005-11-11 at 11:24 +0100, harry gaillac wrote: > Here are some other files. > > Why asterisk send sip OPTION message to agents ? > > Harry > > 2005-11-11 11:2

Re: [Asterisk-Users] Result branching in AEL

"n+101" feature is deprecated and is no longer supported in Asterisk. All applications are modified to set exit status variable. Use something like VoiceMail(b${EXTEN}); if("${VMSTATUS}" = "FAILED") { Noop(mailbox doesn't exists); } On Fri, 2005-11-11 at 10:11 +, Chris Bagnall wrote:

Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)

Do you have silence suppression enabled on your clients? Asterisk can't work with silence suppression. Take a look at http://bugs.digium.com/view.php?id=5374 , the patch works fine to me, now I'm able to set SS and save bandwidth. On Mon, 2005-11-07 at 19:41 -0800, Chris Tracy wrote: I rec

Re: [Asterisk-Users] Caller ID How does it get setup?

AFAIK, most of VOIP providers ignore callerid from ATA and substitute it with a caller id on their records. On Fri, 2005-11-04 at 01:11 -0600, Jason Brashear wrote: OK I am exhausted. I can't seem to figure out how to send a caller ID along with a Outbound call. Can you believe that I go

Re: [Asterisk-Users] Invalid/Timeout handlers in ael?

As far as I (don't:-) understand asterisk sources, extensions 't' and 'i' can be used in _context_ only, but not in macro. Application voicemail has a special handling for 'a' extension to allow it in macro. On Thu, 2005-11-03 at 16:32 -0800, John Biundo wrote: Does anybody know how to code

RE: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana

AFAIK, the official language of this mailing list is English. On Tue, 2005-11-01 at 08:54 -0500, Carlos Alperin wrote: Walter,   No se acerca de que es lo mas atractivo. El servicio puede ser en el horario que tu quieras (Nosotros trabajamos 24 hs, 7 dias a la semana, 36

Re: [Asterisk-Users] Asterisk+Nat+sipura (Help)

I don't think the problem is NAT-related. Looks like "To" header in SIP INVITE message do not match to "User ID" in sipura settings. On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote: Hi ALL;     I have  users with Sipura/Linksys phones regsitered behind Nat(  us

Re: [Asterisk-Users] Format of extensions.conf

Taking in the account poor Asterisk documentation, it's a bug. The bug can be called as a "feature", only when it is documented:-) On Tue, 2005-10-25 at 10:47 -0400, Leif Madsen wrote: > > Set(CALLERID(name)="") > > > > will set the name part of callerid to guess what?-) Yes, to a string > > conta

Re: [Asterisk-Users] Format of extensions.conf

On Tue, 2005-10-25 at 00:52 -0400, Leif Madsen wrote: Now, as someone has also pointed out, using quotes around the string is probably better form as it should handle spaces and such. In expressions only. Set() command is broken in this area (1.2beta and CVS HEAD). To clear, for

Re: [Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf

Check if you have Silence Suppression disabled on PSTN line of spa-3000 (admin/advansed/PSTN line). On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote: I have asterisk running with sipura 3000 connect to PSTN and sipura 2000 connected to phones. On inbound calls I am getting what sounds

Re: [Asterisk-Users] Does fwdout even work anymore?

I just tried to place a call thru fwdout, works fine. On Sat, 2005-10-22 at 06:40 -0700, trixter aka Bret McDanel wrote: Initially I thought this may have been the fiasco last night or the night before (I forget now) where level3 did a software upgrade and it went awry. With the pings respo

RE: [Asterisk-Users] cdr_odbc with tds

...Or fix the problem yourself:-) On Thu, 2005-10-20 at 16:58 +0100, Ben merrills wrote: > What should I do? :) > > Add it to the bug tracker? > > Ben > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Sergey > Okhap

Re: [Asterisk-Users] cdr_odbc with tds

Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a specialist in ODBC, but what seems to me wrong is the module does INSERT into the database, but does not make COMMIT. On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote: > Does anyone know why, using latest cvs head, freetds 0.

Re: [Asterisk-Users] Why Asterisk documentation is so poor...

isk's maintainers roadmap and guidelines. I'm not fluent in english also, to express your wishes in the way you want. On Thu, 2005-10-20 at 15:50 +0200, Olle E. Johansson wrote: > Sergey Okhapkin wrote: > > http://bugs.digium.com/view.php?id=5472 > > > > The users w

[Asterisk-Users] Why Asterisk documentation is so poor...

http://bugs.digium.com/view.php?id=5472 The users will not learn about undocumented AEL features. Sure I'm not going to reopen the problem. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.dig

Re: [Asterisk-Users] goiax configuration help please

Replace [goiax] with [87820]. Just replace the section name. On Wed, 2005-10-19 at 20:21 -0400, Jim Duda wrote: I saw the posting concerning goiax offering free DIDs. I went ahead, created an account, and got myself a DID. Who is goiax, and how can they be doing this for free? It's nice,

Re: [Asterisk-Users] Free DID's

voip.callpacket.com. This is a recent change they made and ser.callpacket.com will not work. But if you nslookup both the names pointing to same ip. They might be using some kind of virtual hosting on that name I think. -Thameem On 10/19/05, Sergey Okhapkin <[EMAIL PROTECTED]>

Re: [Asterisk-Users] Free DID's

On Wed, 2005-10-19 at 14:15 -0700, Jesse Keating wrote: On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote: > Callpacket.com has a free plan (up to 100 mins/month outbound, > unlimited inbound, free DID). Do you have hints on using callpacket w/ Asterisk? register => si

Re: [Asterisk-Users] Free DID's

Callpacket.com has a free plan (up to 100 mins/month outbound, unlimited inbound, free DID). On Wed, 2005-10-19 at 13:31 -0700, trixter aka Bret McDanel wrote: On Wed, 2005-10-19 at 16:19 -0400, Yu Safin wrote: > am I correct in believing that only goiax.com offers free DID's? nope you are

Re: [Asterisk-Users] DID setup from goiax.com

Replace [goiax] with [PHONENUMBER] username= don't work for users in IAX channel. On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote: > That is What I stated in the email.. my GOIAX #. not the DID #. > > That is not the issue. > > > for the incoming context put your goiax.com

Re: [Asterisk-Users] Priority jump in AEL

There is no way in AEL to specify the priority explicitly. To solve the problem use DB_EXISTS function. Here is an example from my dialplan:     if(${DB_EXISTS(Provider/${prov}/used)})     Set(MINUTES_USED=${DB_RESULT}); 

Re: [Asterisk-Users] free dids on goiax.com

I expect all 175 DIDs has gone already... On Tue, 2005-10-18 at 17:58 -0700, John Wenger wrote: On 10/18/05, Matthew Simpson <[EMAIL PROTECTED]> wrote: GoIAX, the Asterisk community's free IAX provider, is offering free US dids now.  I loaded about 175 dids in and

Re: RE:[Asterisk-Users] free dids on goiax.com

I completely agree. No reason to provide unlimited free service, put some reasonable restrictions like no more 10 different numbers could be called a day or no more than 20 calls a day. On Tue, 2005-10-18 at 17:39 -0500, Rajesh kumar wrote: For a free service, its quite acceptable to dema

Re: [Asterisk-Users] sip show peers

Hmm.. What is the output of "sip show users" and "sip show peers"? On Sat, 2005-10-15 at 12:30 -0700, Samy Antoun wrote: --- Sergey Okhapkin <[EMAIL PROTECTED]> wrote: > Are the devices at 200 and 310 set up to register with your asterisk? Yes, they are r

Re: [Asterisk-Users] sip show peers

Are the devices at 200 and 310 set up to register with your asterisk? On Sat, 2005-10-15 at 11:42 -0700, Samy Antoun wrote: Hi, I have 3 SIP extensions, setup as follows: # Device Location options 200 Sipura local 210 Sipura remote nat=yes qualify=yes 310 eyebeam remote nat=yes qua

Re: [Asterisk-Users] Outbound registration expirey

The host you're registering with sets the "expiry" parameter to 60 seconds in the reply message. Use "sip debug" to see SIP messages running. On Fri, 2005-10-14 at 17:00 -0300, Ricardo Poppi wrote: Hi list! I´m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box a

Re: [Asterisk-Users] Starting simple switch from an extension?

DISA(password|context) On Thu, 2005-10-13 at 12:58 +0100, Derek Conniffe wrote: > Hi, > > Is there a command to start "simpleswitch" from an extension? For > example it would allow me to dial in to my * box and get a dial tone to > make an outgoing call. > > Thanks, > > Derek > > __