Re: [Asterisk-Users] v92 modems
[EMAIL PROTECTED] wrote: Hello, Is it possible to use v92 ( a few chipsets version ) modem as FXO PCI modules ? While googling I found some postings on the subject. Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Yes, Modem with an Ambient MD3200, Model # : AMI-IA92/IE92 The Digium products has great performance. Buy Digium products and support Asterisk! Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] FXO port trhoug optimum voice VOIP service
Carlos Trallero wrote: Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p to the phone jack, the dial tone in all of my phones die. Because of the later I'm suspecting that there is some problem with the signaling or voltage detection. My PSTN line is actually from a VoIP modem that runs over the Cablevision network (known as Optimum Voice). Thanks everyone. Carlos __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Where did u get that OEM X100P? Is it MD3200 chip? Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] h323
altus wrote: Good day all Im trying to get asterisk and oh323 to work I following the instruction on http://lists.digium.com/pipermail/asterisk-users/2005- January/081651.html It on fedora core 1,and I downloaded the lated dev. of asterisk Installation: tar -zxvf asterisk-oh323-0.7.1.tar.gz tar -zxvf pwlib-Janus_patch4-src-tar.gz tar -zxvf openh323-Janus_patch4-src-tar.gz cd pwlib ./configure make cd openh323 patch -p1 /root/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch (pach to openh323) cd openh323 ./configure make opt but at make opt I get this error g++: Internal error: Illegal instruction (program cc1plus) Please submit a full bug report. See URL:http://bugzilla.redhat.com/bugzilla for instructions. make[1]: *** [/root/openh323/lib/obj_linux_x86_r/h323rtp.o] Error 1 make[1]: Leaving directory `/root/openh323/src' make: *** [opt] Error 2 Can someone please help Thanks Altus Hello, oh323 how to -- http://linuxpower.blogspot.com/2005/07/h323-supports-for-asterisk.html Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] howto let the stream not passing asterisk
Rosario Pingaro wrote: We need to configure asterisk to authenticate two sip ATAs, but the stream must go directly from one to another ata without tuching asterisk. Is this possible adding canreinvite=yes into sip.conf? is it true laso if asterisk doesn't recognize the spd (t38)? thanks Rosario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Yes, If they support the same codec and don't put t / T with Dial command on d extensions.conf. ATA186 has a problem with canreinvite=yes for more info http://lists.digium.com/pipermail/asterisk-doc/2004-June/000547.html Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Kumara, Yes, Without problems. Can u install RH9 on ur box? Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
Zachary Whitley wrote: On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote: Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Kumara, Yes, Without problems. Can u install RH9 on ur box? Cheers, ~Madhawa I'm assuming that Madhawa is suggesting that you install RH9. I've installed Asterisk on FC4 with very few problems. Start with a standard FC4 installation then install the following rpms from atrpms.net: Hi Zachary! I'm not suggesting installing RH9 :) I mean, whether he can install RH9 on same box without any problems. He said he couldn't install @least FC4 on the box, do you have any idea? problems of IRQ sharing with the card? Best regards, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco ATA and a PayPhone
Mark Johnson wrote: Hi Mark, I think ATA-188 supports polarity reversal. Cheers, ~Madhawa I hope I don't sound stupid, but what does that mean? I can't find a definition for polarity reversal and how it would help me. I do see the 188 supports it, but I'm not sure what to do with it. Thanks!! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Mark, I've done this using SPA-2000, SPA-2000 can generate polarity reversal signal, The pay-phone detects call answer and hangup by revesal signal. also the pay-phone must be supported polarity reversal detection. http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800a6210.shtml http://michigantelephone.mi.org/distribute.html Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] How scalable is asterisk
Alex Ponnath wrote: Hi, I am new to Asterisk and I am wondering if anyone has some real live Data on How well Asterisk scales ? I am interested on how many TDM to Voip and How many Voip to Voip calls it can handle. I am also wondering if anyone has used Asterisk more like a softswitch then A PBX to provide Services to End users. Thanks for any Answers and or pointers in Advance Alex ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Alex, Please find answers bellow, Q. I am interested on how many TDM to Voip and How many Voip to Voip calls it can handle. A. we've tested 120 calls with Dual Xeon 2.8GHz /3GB Ram Q. How many Voip to Voip calls A. our Asterisk box supports upto 360 IP 2 IP calls, without transcoding. Q.I am also wondering if anyone has used Asterisk more like a softswitch then A PBX to provide Services to End users. A. Yes, we're using Asterisk as a soft switch. Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Full T38 sip Faxing now Available
Kanuri, Seshu (Company IT) wrote: Michael, Here are some of the reactions to your original post on the T38 FAX thingy: 1) Why the big secret? Why not post your solution to the list? 2) It's probably just another one of those nasty closed source add-ons for sale. 3) I'm guessing it has nothing to do with *. Probably MAX TNT, Cisco, or some other soft-switch. In other words, nothing new. You have not answered any of those questions, but here you are accusing someone as un-professional. Why don't you answer their questions first and then we all can decide if your post was a spam or not. Seshu Seshu, I agree with u. Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco ATA and a PayPhone
Mark Johnson wrote: I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones I think). If you can put the money in and dial real fast, it works!! I have been screwing with the ATA configs for days now and can't come up with a way to extend the timeout or to even disable it. Anyone have any suggestions or could recommend another method? FX ports may be an option, but they are pretty far from where these phones are going to go. As always, thanks for any input!! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Mark, Sipura SPA-2000 supports polarity reversal. Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco ATA and a PayPhone
Mark Johnson wrote: I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones I think). If you can put the money in and dial real fast, it works!! I have been screwing with the ATA configs for days now and can't come up with a way to extend the timeout or to even disable it. Anyone have any suggestions or could recommend another method? FX ports may be an option, but they are pretty far from where these phones are going to go. As always, thanks for any input!! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Mark, I think ATA-188 supports polarity reversal. Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] List
Huddleston, Robert wrote: Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello list, We've same problem. ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] problems with compiling asterisk-oh323
wassim darwish wrote:
i ve downloaded
asterisk-oh323-0.6.6.tar.gz
I am getting this and anybody know howto fix this?
#tar zxvf asterisk-oh323-0.6.6.tar.gz
oh323]# cd asterisk-oh323-0.6.6
asterisk-oh323-0.6.6]# ls
asterisk-driver CONFIGURATION Makefile rpm
TESTS
BUGS COPYINGREADMErules.mak
wrapper
asterisk-oh323-0.6.6]# make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
./check_ver /root/src/oh323/pwlib pwlib
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
./check_ver /root/src/oh323/openh323 openh323
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
g++ -Wall -x c++ -Os -DUSE_OLD_CAPABILITIES_API=1
-DWRAPTRACING -DWRAPTRACING_LEVEL=5
-DPWLIBVERSION=\..\ -DOPENH323VERSION=\..\
-I/root/src/oh323/pwlib/include
-I/root/src/oh323/openh323/include
-I/root/src/oh323/openh323/include/openh323
-I../asterisk-driver -c wrapper_misc.cxx -o
wrapper_misc.o
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:35:19: ptlib.h: No such file or
directory
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:61: error: expected class-name before
'{' token
wrapper_misc.hxx:63: error: `PMutex' has not been
declared
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `PCLASSINFO' with no type
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `parameter' with no type
wrapper_misc.hxx:68: error: `BOOL' does not name a
type
wrapper_misc.hxx:73: error: `PString' does not name a
type
wrapper_misc.cxx: In constructor
`WrapMutex::WrapMutex(char*)':
wrapper_misc.cxx:48: error: class `WrapMutex' does not
have any field named `PMutex'
wrapper_misc.cxx:50: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:50: error: (Each undeclared
identifier is reported only once for each function it
appears in.)
wrapper_misc.cxx:50: error: `PString' undeclared
(first use this function)
wrapper_misc.cxx:51: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:51: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:51: error: `endl' undeclared (first
use this function)
wrapper_misc.cxx: At global scope:
wrapper_misc.cxx:54: error: `BOOL' does not name a
type
wrapper_misc.cxx: In member function `void
WrapMutex::Signal(const char*, int, const char*)':
wrapper_misc.cxx:78: error: `PMutex' has not been
declared
wrapper_misc.cxx:78: error: no matching function for
call to `WrapMutex::Signal()'
wrapper_misc.cxx:77: note: candidates are: void
WrapMutex::Signal(const char*, int, const char*)
wrapper_misc.cxx:79: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:79: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:79: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:79: error: `endl' undeclared (first
use this function)
make[1]: *** [wrapper_misc.o] Error 1
make[1]: Leaving directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
make: *** [subdirs_build] Error 1
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Hello wassim,
First of all you
Re: [Asterisk-Users] does h323 exists in astcc trunks
wassim darwish wrote: does astcc support h323 ,because it doesnt exists h323 in trunks technology. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Wassim, Yes, ASTCC supports H323 trunk, but u have to install channel H323 or OH323 on * box. you can download oh323 source from http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.6.tar.gz Cheers, ~Madhawa Blog http://linuxpower.blogspot.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Configuration
Afzaal Mirza wrote: Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, You can find basic information from my blog http://linuxpower.blogspot.com I'm making a visual guide using flash and next week I'll post on my blog, if you have a question , ask me. Asterisk basic configuration http://linuxpower.blogspot.com/2005/07/asterisk-basic-configurations.html Cheers, ~Madhawa Blog http://linuxpower.blogspot.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 VoIP provider
Bernie Courtney wrote: looking at setting up an asterisk box at my home-- what VOIP providers are you all using with the best results (and low costs! lol) thanks Bernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Bernie, 1) best results www.nufone.net 2) low cost www.voipjet.com Cheers, Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problem while capturing DTMF digits in AGI
Hello,
You must answer the channel
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $tests = 0;
my $pass = 0;
my $fail = 0;
#setup callback
$AGI-setcallback(\mycallback);
print STDERR AGI Environment Dump:\n;
foreach $i (sort keys %input) {
print STDERR -- $i = $input{$i}\n;
}
$AGI-answer();#---Answer the channel
Cheers,
Madhawa
somesh s wrote:
Hi All,
I have some problem while capturing DTMF digits in AGI
script.
My configuration for user is ..
[9009]
type=friend
host=dynamic
context=default
dtmfmode=rfc2833
dtmfrelax=no
disallow=all
allow=ulaw
allow=h263
canreinvite=yes
[9010]
type=friend
host=dynamic
context=default
dtmfmode=rfc2833
dtmfrelax=no
disallow=all
allow=ulaw
allow=h263
canreinvite=yes
And the script read like ...
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $tests = 0;
my $pass = 0;
my $fail = 0;
#setup callback
$AGI-setcallback(\mycallback);
print STDERR AGI Environment Dump:\n;
foreach $i (sort keys %input) {
print STDERR -- $i = $input{$i}\n;
}
my $timeout = 3000; # 3 second timeout
my $wait = 1; # true
my $outgoing_number;
while($wait) {
my $digit = $AGI-wait_for_digit($timeout);
print STDERR Digit ($digit)\n;
if ($digit 0) {
$digit -= 48;
$outgoing_number .= $digit;
}
else { #stop waiting for more digits
$wait = 0;
}
}
print STDERR Outgoing number ($outgoing_number)\n;
--
I will get the output as Outgoing number () and
digits(0)
Am I missing something here? Please do help me in this
regard.
Regards
Somesh S Shanbhag
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