Re: [Asterisk-Users] Another software like Asterisk?
Try Vovida's Vocal, i think it does it. Mireia Munoz de jesus wrote: Hi! I am looking for a software that can work as h.323 - sip gateway other than asterisk and free. Someone can help me? Thanks. Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk at the beginning
Thomas Schroeter wrote: Hi, I just got started with Asterisk. Installation was OK, no errors. But how do I activate the IAX and SIP channels now? I loaded the modules, but nothing happened, there's no connection to the relavant ports. Anything I forgot to do...? >> Yes, you forgot to read the documentation. Read it and ask again. Read the wiki-tiki too, http://www.voip-info.org/wiki-Asterisk You need to not only activate the channels, but mess up with their config files and define peers/friends/clients. Thanks in advance! thomas --- Thomas Schroeter // +49-175-4624147 // +49-40-72976451 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'Busy tone' after hangup
Yes that's right, it's the TDM400P which generates it. Ryan Courtnage wrote: On 29-Mar-04, at 3:23 PM, NetOne Administrator wrote: As you see, * generates no busy tone, it hangs up the channel. It's your client which generates the tone. This is not something to be done from *. Thanks, So in the case of my cheap analog phones, would the 'client' be the phones themselves, or my TDM400P? I'm _guessing_ the TDM400P, since this 'busy tone' sounds identical on every analog phone I plug into it...? Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'Busy tone' after hangup
As you see, * generates no busy tone, it hangs up the channel. It's your
client which generates the tone. This is not something to be done from *.
Regards,
Doichin Dokov
Ryan Courtnage wrote:
Hello,
I find that when 2 extensions are connected, and one of the extensions
hangs up on the call, the other will receive a busy signal (as if to
indicate that the call is over).
Does this sound like a config problem, or is it the default behavior
of *?
Example:
[ext-testing]
exten => 111,1,Dial(SIP/2001)
exten => 111,2,Hangup
exten => h,1,Hangup
Zap/2 dials 111:
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial("Zap/2-1", "SIP/2001") in new stack
-- Called 2001
-- SIP/2001-b164 is ringing
-- SIP/2001-b164 answered Zap/2-1
After SIP/2001 hangs up:
-- Executing Hangup("Zap/2-1", "") in new stack
-- Hungup 'Zap/2-1'
... followed by Zap/2 getting a "beep-beep-beep-..." 'busy tone'.
If this is the default behavior, can it be changed? After the remote
end hangs-up on a call, I'd expect to hear either dialtone or silence.
Thanks
Ryan
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Re: [Asterisk-Users] windows alternitives to Asterisk?
[EMAIL PROTECTED] wrote: SSH >> Nice :))) On Mon, 8 Mar 2004, hank smith wrote: is there a program that I can install on my linux box so I can configure the pbx from the internet from my windows box so I don't have to work with config files? thanks hank - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 08, 2004 11:07 AM Subject: Re: [Asterisk-Users] windows alternitives to Asterisk? hank smith wrote: hello I am just curious if there is any windows alternitives to Asterisk? can I also use them with free world dialup? thanks hank No, but maybe you could port Asterisk to Windows. No, that's not a joke. The Zaptel drivers might be tough, but Asterisk's VoIP features would probably run under Cygwin without too high a mountain of work. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Options for 3+ FXO ports
Our telco here in Bulgaria has this option - you can get what number of lines you want on a primary (E1 here). You choose how much lines - from 1 to 30 - go to the primary. Price is according to number of channels, but signalling is primary! Check out at your telco if this is not possible out there (I believe it should) Jorge Mendoza wrote: [EMAIL PROTECTED] wrote: Original Message Subject: Re: [Asterisk-Users] Options for 3+ FXO ports From: "Jorge Mendoza" <[EMAIL PROTECTED]> Date: Mon, March 08, 2004 8:53 am To: [EMAIL PROTECTED] Rich Adamson wrote: I'm looking into implementing an * solution and I'm expecting to require 4 incoming FXO lines. Based on my reading of the mailing list archives and what I read on the Digium web site, I don't think that this can be achieved with X100P cards as it doesn't seem to be practical to have more than 3 X100P cards at the time (IRQs is the main problem I think). At four FXO ports, I can't quite justify something like a full T1 on a monthly basis. But, since I've never ordered one, I'm thinking that it's a "all or nothing" deal (i.e.: I can't order a T1 and pay for only 4 voice channels). It would seem that the best I can do is a 3x8 configuration according to the information on this page The Mediatrix 1204 FXO gateway sort of works with asterisk in the US, but have had a few issues with CallerID being recognized when the four pstn lines have different ring cadence (at least I believe its a cadence issue). Other then that, there seems to be a significant market opportunity for anyone that can produce a reliable 4-port device. A goggle search for channel banks turned up www.nextag.com with samples such as this: Adtran TA750 Chassis w/psu ac adapter $900 Adtran Quad FXO card: $230 Adtran TA750 w/12 FXS ports: $1250 Adtran TA750 w/FT1, 20 FXS: $1900 However, I don't consider those prices cost effective for four ports after adding in the digium T1 card, etc. I'm interested in a 4-port as well if anyone knows of something. See: http://www.welltech.com.tw/ No yet tested, but waiting my first samples next week. Have callerid and reversal polarity detection!. Prices seems to be half of Mediatrix. Jorge I have been eagerly awaiting word on these devices! They would not give me one on evaluation, and charge a 20% restocking so I have not tried them myself. Please post your findings :-) Michael Graves I will Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.
CW_ASN wrote: Are you Klaus? If you're not Klaus, you wrote another patch. If you're Klaus, as you see, works in that way. >> Nopez i'm not In that case, exists another patch from a guy called Klaus. I'm using this patch since Dec2003. Maybe helps, I don't know, but this is other alternative. Its merged with the last app_dial from CVS, maybe isn't correct for the last status (announce override). Well, i don't know what Klaus did, what I see in the patch you attached is what I did - and that it what was merged some days ago in CVS. Please have a look at bug #1107. My nick is lancey, have a look there. Best regards, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.
Soren Rathje wrote: "Hans-Henrik Andresen" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] HMM - This wont work :( exten => 10,1,Dial(SIP/hha1,20,S(10)) exten => 10,2,VoiceMail,u10 exten => 10,102,VoiceMail,b10 When did you checkout your version of Asterisk from CVS ?? This "feature" was put into CVS on the 6'th as a fix for bug #1107 but I have not seen it in "v1-0_stable". This is a new feature, that's why it is NOT in 1.0-stable. Only bugfixes go into -stable. New feautres - in CVS. -- Søren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.
CW_ASN wrote: >> This is wrongs. It's me who wrote the patch, it's available in CVS Are you Klaus? If you're not Klaus, you wrote another patch. If you're Klaus, as you see, works in that way. >> Nopez i'm not ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.
CW_ASN wrote: People: 1) Some guy wrote a app_dial modification to start to count time when answer arrives. Interested? (thanks to Luciano!) AbsoluteTimeOut counts time since this statement is executed... If you have a long ring time (without answer), it is counted! With the proper patch, the time is counted since answer is recognized. The sintax is: [Description]: Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL ][|whenhangup]): whenhangup is the time to hangup the call in seconds. This time is counted since answer is detected. This parameter is a bit different than AbsoluteTimeOut >> This is wrongs. It's me who wrote the patch, it's available in CVS also. See 'help application dial' for the correct options. It's an addition to Dial cmd transfer options. It's set via S(number-of-seconds). So try something like this: Dial(tech/channel,timeout,S(xx)) This is the way it works, for sure. 2) Other guy wrote a patch for channel.c to insert tones "x" seconds before hangup. I can't remember his name (Nicolás, from Rosario?), please look in http://bugs.digium.com/ Hope this helps, Gus - Original Message - From: "Senad Jordanovic" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, March 07, 2004 3:26 PM Subject: RE: [Asterisk-Users] Re: Re: Limit on call in minuttes. Sorry, but that IS NOT implemented into asterisk... (as far I know of). Hi, This isn't quit good :( The caller have the message played, but the called person are cut off without any warning.. I hoped to be warned, like "In 1 minnute the line will be disconected", or just som beep beep /HHA "Senad Jordanovic" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] Yes, it is. (If I remember correctly :) It is "T" that you need to include in that context. [$CONTEXT] exten => 1,AbsoluteTimeout($SECONDS) exten => 2,Dial($SOMETHING) exten => T,Playback($YOURMESSAGE) Save $YOURMASSAGE in /var/lib/asterisk/sounds If above does not work, please let me know. Ta SJ Thank you This works, but. It just cut the line, I had hoped for some bip bip bip to remind that now your about to be disconected, is this possible as well ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA 186 Firmware
v.3.0 works fine too James Coberly wrote: v2.16.2 ata18x Works Fine for me. - Original Message - From: "Erick Weber V." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 03, 2004 1:18 PM Subject: [Asterisk-Users] Best ATA 186 Firmware Hi: Someone know wich is the best firmware for the ATA 186 with * Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !
FreeBSD asterisk port is *NUTS* Don't use it! Asterisk compiles just fine on BSD, if you are using 4.x-RELEASE, and not using chan_h323, chan_oss, zaptel & libpri. Darren Wiebe wrote: Sorry to just come on line now. Have you tried the FreeBSD port? net/asterisk is the place to look. It always dumps core on me but you may have better luck. Darren Wiebe [EMAIL PROTECTED] William Waites wrote: On Mon, Mar 01, 2004 at 11:37:59AM +0100, Serge wrote: Thanks William, it's get. but new problem: server dont have any sound device ( I think:) ) Why noone make normal Makefile and FAQ for FreeBSD Asterisk.. als many for Linux. I don't know why the Asterisk crowd is resistant to using GNU Autoconf, it solves these problems very neatly. OSS doesn't work on FreeBSD, just erase chan_oss from the Makefile in the drivers directory... /w ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stable how to compile ?
Kelly Murphy wrote: Follow all the instructions on http://www.asterisk.org/index.php?menu=download You still need to checkout libpri and zaptel. If you want more information checkout http://www.voip-info.org This is the main repository if information on Asterisk. >> No you don't need to checkout libpri and zaptel if not using Digium hardware. Asterisk will compile fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SamW Sent: Monday, March 01, 2004 10:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk stable how to compile ? I want to build a stable asterisk to run, if some one can guide through how to compile will be useful. Currently available documentation do not show any good information about a correct how to. According to the Asterisk Web site, it indicate to download the Stable 1.0 use the following, cvs checkout -r v1-0_stable asterisk. But Asterisk won't build on its own, it needs libpri and zaptel. There are 2 places to download libpri and zaptel, 1. CVS cvs checkout libpri zaptel 2. Use Download site, following 2 locations, ftp://ftp.asterisk.org/pub/telephony/libpri/libpri-0.5.2.tar.gz ftp://ftp.asterisk.org/pub/telephony/zaptel/zaptel-0.8.1.tar.gz Which one of the 2 above should be used for a stable Asterisk build. (I do not use digium hardware) I am currently seeing lot of segmentation faults (core-dump) when I running asterisk. Help is appreciated. - SamW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
Angel Gabriel wrote: I have 5 BT phone lines coming into my office. We use four for international calls, and one for local/mobile calls. We have just obtained another call carrier, and now we would like to be able to make calls from any phone to any carrier, without having to remember what details to tap into the phone. I would like all calls to be prefixed with the relevant codes, so that my employees can all dial direct. Also, incoming calls, I want them all redirected to just one phone, the one in reception, and then diverted as required. Is the above possible?? Yes it is. Digium cards and asterisk setup. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stable how to compile ?
If you are not using Digium hardware, then you don't need libpri and zaptel. Asterisk WILL build on its own. SamW wrote: I want to build a stable asterisk to run, if some one can guide through how to compile will be useful. Currently available documentation do not show any good information about a correct how to. According to the Asterisk Web site, it indicate to download the Stable 1.0 use the following, cvs checkout -r v1-0_stable asterisk. >> But Asterisk won't build on its own, it needs libpri and zaptel. >> Absolutely wrong There are 2 places to download libpri and zaptel, 1. CVS cvs checkout libpri zaptel 2. Use Download site, following 2 locations, ftp://ftp.asterisk.org/pub/telephony/libpri/libpri-0.5.2.tar.gz ftp://ftp.asterisk.org/pub/telephony/zaptel/zaptel-0.8.1.tar.gz Which one of the 2 above should be used for a stable Asterisk build. (I do not use digium hardware) I am currently seeing lot of segmentation faults (core-dump) when I running asterisk. Help is appreciated. - SamW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP developer Wanted ! :-)
What is your * app? What should the frontend do? Greetings, Doichin Dokov reseaux wrote: Dear ALL i need to develop a web frontend for my * app i need only manage data from MySQL db, i will pay to develop it (not much :-) ) Thanks in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk on FreeBSD.
Hello! Chan_H323 is not working under FreeBSD, at the moment. Dmitry Mishchenko wrote: Hello! I've got the latest asterisk sources from cvs dated Jan 26 2004. And was trying to build it under FreeBSD 4.8 Main part was built fine. I face problems only while compiling h323 channel. Its Makefile has hardcoded linux libs (openh323 and pwlib) and some defines. Also -lpthread was replaced with -pthread and -ldl with -dl. So I've got successfully build and installed asterisk. Now major problems comes: After starting asterisk it is trying to get all available CPU time. I'm using standard config files. Turning off modules h323 and oss didn't help. System looks working, I can dial test extension with SIP from other host and navigate "demo". Any ideas why it takes so much resources? Dmitry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan h323 Compile problem
Do not compile openh323 and pwlib from cvs.
Use the versions described in the README of chan_h323 so (in
channels/h323 dir).
Good luck!
Doichin Dokov
Mike Bentley wrote:
Hi can anyone help me with this
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC
-Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT
-D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
-DPTRACING -DP_USE_PRAGMA -I../../include
-I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include
-I/usr/src/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h323.cpp
In file included from /usr/src/pwlib/include/ptlib.h:169,
from ast_h323.h:30,
from ast_h323.cpp:29:
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: parse error
before `
protected'
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: syntax error
before `*'
token
In file included from /usr/src/pwlib/include/ptlib.h:181,
from ast_h323.h:30,
from ast_h323.cpp:29:
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:53: parse error
before `public
'
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:55: destructors must
be member
functions
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:57: parse error before `
protected'
In file included from /usr/src/pwlib/include/ptlib.h:187,
from ast_h323.h:30,
from ast_h323.cpp:29:
/usr/src/pwlib/include/ptlib/args.h:121: parse error before `{' token
/usr/src/pwlib/include/ptlib/args.h:147: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:156: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:165: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:175: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:190: `ostream' was not declared in
this
scope
/usr/src/pwlib/include/ptlib/args.h:191: `strm' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:191: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:191: variable or field `PrintOn'
declared
void
/usr/src/pwlib/include/ptlib/args.h:197: `istream' was not declared in
this
scope
/usr/src/pwlib/include/ptlib/args.h:198: `strm' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:198: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:198: variable or field `ReadFrom'
declared
void
/usr/src/pwlib/include/ptlib/args.h:206: parse error before `&' token
/usr/src/pwlib/include/ptlib/args.h:215: parse error before `&' token
/usr/src/pwlib/include/ptlib/args.h:246: virtual outside class
declaration
open h323 and pwlib from cvs complied fine.
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[Asterisk-Users] G.729 quiestion
Hi all! If i purchase the G.729 codec for *, can Asterisk use it for convertion, or just pass-through only? I need to be able to convert from G.729 to iLBC (or GSM maybe) and vice versa. Is it possible with *? Greetings, Doichin Dokov NetOne - Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?
Do you mean I would not be able to run H323, or Asterisk as a whole? Because everything else (SIP-to-SIP calls, Messages, VoiceMail, and other apps) seem to be working just perfect. I am now considering using some other product as a SIP/H323 gateway, and still use Asterisk. I really do not want to run a Linux machine for this, I'm not so experienced in linux as in FreeBSD, and do not think it would work more stable than a BSD box. I would also like to run another things on that machine, most of which are BSD-specific (VPNs our company uses, and so on). I would also like to have a good QoS control, which I now have on BSD thanks to ALTQ queing. Any suggestions & comments? Doichin Dokov NetOne - Silistra, Bulgaria Tilghman Lesher wrote: On Sunday 11 January 2004 16:18, NetOne Administrator wrote: I'm trying to set up Asterisk on FreeBSD 4.9 to route calls to H.323 GK. I have installed asterisk using the ports. It seems to be running OK, but when i try to dial through h323, it segfaults I want you to look at the headers of my reply and note that I'm running my mail client on FreeBSD. Now my advice: run your Asterisk server on Linux. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on FreeBSD 4.9?
Hi all I'm trying to set up Asterisk on FreeBSD 4.9 to route calls to H.323 GK. I have installed asterisk using the ports. It seems to be running OK, but when i try to dial through h323, it segfaults I'm using X-Lite as SIP client, i have set up my h323.conf [general port = 1721 bindaddr = 0.0.0.0 tos = lowdelaydtmfmode = rfc2833 context = Out noFastStart = yes noH245Tunneling = no gatekeeper = 01.23.45.67 AllowGKRouted = no allow = all [NET1-BG] type=h323 prefix=0 context=Inc and in my extensions.conf there's a line like this: exten = _0.,1,Dial(H323/0889811777,20) which i think should dial the phone entered, no matter what number is dialed by the client (if it start with 0, of course). Anyone with suggestion? where am i wrong? Doichin Dokov NetOne - Silistra 0889 / 811-777 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on FreeBSD 4.9
Hi all!I'm trying to set up Asterisk on FreeBSD 4.9 to route calls to H.323 GK.I have installed asterisk using the ports.It seems to be running OK, but when i try to dial through h323, it segfaults.I'm using X-Lite as SIP client, i have set up my h323.conf:[general]port = 1721bindaddr = 0.0.0.0tos = lowdelaydtmfmode = rfc2833context = OutnoFastStart = yesnoH245Tunneling = nogatekeeper = 01.23.45.67AllowGKRouted = noallow = all [NET1-BG]type=h323prefix=0context=Incand in my extensions.conf there's a line like this:exten => _0.,1,Dial(H323/0889811777,20) which i think should dial the phone entered, no matter what number is dialed by the client (if it start with 0, of course).Anyone with suggestion? where am i wrong?Doichin DokovNetOne - Silistra0889 / 811-777 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
