Re: [Asterisk-Users] festival and gcc 3.3.2 (Fedora Core 1)
That did it.
thank you, thank you, thank you,
-reed
Marc Sutter wrote:
Hi,
had the same problem... and we wrote a patch.
This patch's are for speech_tools 1.2.3 and festival 1.4.3.
to use in the corresponding directory with:
#patch -p1
Hope this help. If so let it know.
Have fun !!!
On Sat, 2004-05-01 at 02:35, Reed Wade wrote:
Can someone tell me how to build festival on a machine with gcc 3.3.2?
I've searched all around and even found a reference or two that the
problem exists but I'm not seeing the fix.
thanks!
-reed
Symtoms are --
./configure, then
[EMAIL PROTECTED] speech_tools]# make
Check system type
Remake modincludes.inc
NATIVE_AUDIO
ok
EDITLINE
config/modules/editline.mak
SIOD
siod/siod.mak
WAGON
stats/wagon/wagon.mak
SCFG
grammar/scfg/scfg.mak
WFST
grammar/wfst/wfst.mak
OLS
stats/ols.mak
RXP
rxp/rxp.mak
LINUX16_AUDIO
config/modules/linux16_audio.mak
Making in directory ./siod ...
making dependencies -- siodeditline.c cc1: warning:
"-Wno-non-template-friend" is valid for C++ but not for C/ObjC
cc1: warning: "-Wno-deprecated" is valid for C++ but not for C/ObjC
el_complete.c cc1: warning: "-Wno-non-template-friend" is valid for C++
but not for C/ObjC
cc1: warning: "-Wno-deprecated" is valid for C++ but not for C/ObjC
editline.c cc1: warning: "-Wno-non-template-friend" is valid for C++ but
not for C/ObjC
cc1: warning: "-Wno-deprecated" is valid for C++ but not for C/ObjC
el_sys_unix.c cc1: warning: "-Wno-non-template-friend" is valid for C++
but not for C/ObjC
cc1: warning: "-Wno-deprecated" is valid for C++ but not for C/ObjC
slib.cc slib_core.cc slib_doc.cc slib_file.cc slib_format.cc
slib_list.cc slib_math.cc slib_sys.cc slib_server.cc slib_str.cc
slib_xtr.cc slib_repl.cc siod_fringe.cc siod_server.cc io.cc trace.cc
EST_SiodServer.cc siod.cc siod_est.cc
g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend
-Wno-deprecated -DSUPPORT_EDITLINE -I../include slib.cc
In file included from ../include/EST_String.h:50,
from ../include/siod.h:17,
from slib.cc:88:
../include/EST_iostream.h:54:26: strstream.h: No such file or directory
make[1]: *** [slib.o] Error 1
make: *** [siod] Error 2
[EMAIL PROTECTED] speech_tools]#
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diff -ur festival.bad/src/modules/base/phrasify.cc
festival/src/modules/base/phrasify.cc
--- festival.bad/src/modules/base/phrasify.cc 2001-04-04 13:55:20.0 +0200
+++ festival/src/modules/base/phrasify.cc 2004-04-25 15:57:52.0 +0200
@@ -218,7 +218,7 @@
EST_Val npbreak = wagon_predict(w,phrase_type_tree);
w->set("pbreak",npbreak.string()); // may reset to BB
}
- pbreak = w->f("pbreak");
+ pbreak = w->f("pbreak").string();
if (pbreak == "B")
w->set("blevel",3);
else if (pbreak == "mB")
diff -ur festival.bad/src/modules/base/word.cc festival/src/modules/base/word.cc
--- festival.bad/src/modules/base/word.cc 2001-04-04 13:55:20.0 +0200
+++ festival/src/modules/base/word.cc 2004-04-25 15:59:55.0 +0200
@@ -64,10 +64,10 @@
for (w=u->relation("Word")->first(); w != 0; w = next(w))
{
lpos = NIL;
- pos = ffeature(w,"hg_pos");
+ pos = ffeature(w,"hg_pos").string();
// explicit homograph pos disambiguation
if (pos == "0")
- pos = ffeature(w,"pos");
+ pos = ffeature(w,"pos").string();
if (pos != "0")
lpos = rintern(pos);
@@ -100,8 +100,8 @@
// from which a list can be read.
EST_String p;
-if (((p = ffeature(w,"phonemes")) != "0") ||
- ((p = ffeature(w,"R:Token.parent.phonemes")) != "0"))
+if (((p = ffeature(w,"phonemes").string()) != "0") ||
+ ((p = ffeature(w,"R:Token.parent.phonemes").string()) != "0"))
{
LISP phones = read_from_lstring(strintern(p));
diff -ur festival.bad/src/modules/Intonation/int_tree.cc
festival/src/modules/Intonation/int_tree.cc
--- festival.bad/src/modules/Intonation/int_tree.cc 2001-04-04 13:55:20.0
+0200
+++ festival/src/modules/Intonation/int_tree.cc 2004-04-25 15:58:42.0 +0200
@@ -87,11 +87,11 @@
for (s=u->relation("Syllable")->first(); s != 0; s=next(s))
{
if ((paccent = accent_specified(s)) == "0") // check if pre-specified
- paccent = wagon_predict(s,accent_tree);
+ paccent = wagon_predict(s,accent_tree).string();
if (paccent != "NONE")
[Asterisk-Users] phonejack and linejack in the same system
Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 01:16:42 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 01:16:43 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! this does not happend and asterisk runs ok and gives a CLI> prompt if i comment back these lines. Any hint? Regards, -=Raul=- ___ 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] phonejack and linejack in the same system
I'm a newbie too -- search the archives for "ztdummy". -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raul Elizondo (wizardteam) Sent: Sunday, May 02, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] phonejack and linejack in the same system Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 01:16:42 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 01:16:43 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! this does not happend and asterisk runs ok and gives a CLI> prompt if i comment back these lines. Any hint? Regards, -=Raul=- ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
Hi, > -Original Message- > On Sat, 2004-05-01 at 16:42, Gavin Hamill wrote: > > Ah I've only noticed this thread has forked to CLID-in-non-Bellcore > > areas :) PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE can some > > Kind and Worthy soul spend a little time in getting this really > > important feature implemented? You would have the undying > gratitude of > > thousands of X100P users all round the world! :D Without CallerID > > support, the amount of 'cool stuff' you can do on a 1 line > system is > > much reduced! :( > > (please?) > > How about setting up a bounty? > http://voip-info.org/wiki-Asterisk+bounty Yes: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+non-Bellcore-CLID Florian ___ 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] iconnecthere behind NAT, strange deal
I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my "plan" with them. Now all my calls get intercepted immediately, "We're sorry, but your account is temporarily unavailable." Incoming calls work just fine. I contacted their so-called "customer care," which has sent me repeated replies asking me to give them the version of my PC phone. When I say I don't have one, they say, "Sorry, we only help those who do." I like to play with their GSM stuff, so I hate to let the account go, but if no one here knows what might be going on, they certainly don't. FWIW I used to prepend "" to the dialed number, and it worked fine until last week. Thx. B. ___ 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] phonejack and linejack in the same system
i actually found a couple things interesting in the archives after i wrote my first email, and i deleted the soundcard, but i still got problems with the linejack, i m testing all possible options. -=Raul=- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Sunday, May 02, 2004 1:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] phonejack and linejack in the same system I'm a newbie too -- search the archives for "ztdummy". -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raul Elizondo (wizardteam) Sent: Sunday, May 02, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] phonejack and linejack in the same system Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 01:16:42 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 01:16:43 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! this does not happend and asterisk runs ok and gives a CLI> prompt if i comment back these lines. Any hint? Regards, -=Raul=- ___ 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] phonejack and linejack in the same system
This is as far as i could go... I readed about the archives for "ztdummy", and made the fix and recompiled after using ./configure --disable-isa-pnp, i edited the phone.conf and extensions.conf, but i still get this messages: [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 03:52:44 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 03:52:44 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 03:52:44 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 03:52:44 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! [EMAIL PROTECTED] asterisk]# ls /dev/phone0 /dev/phone0 I am going to compile openh323 just to make sure both quicknet cards work in this box, but it was working in another box with redhat9. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Raul Elizondo (wizardteam) Sent: Sunday, May 02, 2004 3:28 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] phonejack and linejack in the same system i actually found a couple things interesting in the archives after i wrote my first email, and i deleted the soundcard, but i still got problems with the linejack, i m testing all possible options. -=Raul=- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Sunday, May 02, 2004 1:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] phonejack and linejack in the same system I'm a newbie too -- search the archives for "ztdummy". -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raul Elizondo (wizardteam) Sent: Sunday, May 02, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] phonejack and linejack in the same system Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 01:16:42 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 01:16:43 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! this does not happend and asterisk runs ok and gives a CLI> prompt if i comment back these lines. Any hint? Regards, -=Raul=- ___ 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]
[Asterisk-Users] RE: Caller ID
Tracy R Reed [EMAIL PROTECTED] wrote: > > > > > > How about setting up a bounty? > > > http://voip-info.org/wiki-Asterisk+bounty > > > > > If I had the money to rent-a-coder, would I have begged on a public > > mailing list? > > > You are missing the point. If you contribute $20 and everyone else who > wants this feature puts up their $20 it eventually adds up to enough money > to get someone interested. You don't have to bear the entire burden > yourself. > I'm not sure if software support for the line reversal monitoring is desirable in the driver. As the bounty calls for the code to be accepted into CVS, this would be an issue for anyone looking to collect. There were rumours, all those years ago :-), that the new FXO modules would support line reversal detection in hardware. I can't find any technical specifications or other documentation for the new modules so I can't tell if the modules are aware of a world outside of the USA. I hope that they have this capability. I also hope that they have the relevant approvals for connection to the PSTN etc. I was rather hoping that, by now, one of the $15 X100P manufacturers would have released a $16 X100P card, with a slightly upgraded chipset, that was suitable for international operation. Digium seem to have more of a profit margin with their $100 version, so they could probably release an updated card (X102P?) without it affecting the price. At the moment, the choices seem to be: 1. Throw the X101P in the bin and spend $133 on a TDM01B (if it does turn out to support non-USA Caller*ID). 2. Wait for a $16 X100P clone, with an upgraded chipset, and throw the Digium X101P in the bin. 3. Get the soldering iron out and hack serial port support into a BT CD50. 4. Get a digital line (ISDN etc.) Options 1 and 2 still rely upon driver support, but I'd expect the driver changes to be identical, or very similar, no matter which option was taken. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Caller ID
Wake up. The reversal detection is a complete waste of time. Totally unnecessary. Pointless. A line break detector would have much more use, as it would give a reliable disconnect detection on many lines. (Actually, reversal detection would have years ago, but its not much use any more). All you need for these CLI requirements is to monitor for some energy on the line. Since these FXOs are not being used in banks of hundreds, you will never notice this MIPs this uses. Regards, Steve Kevin Walsh wrote: Tracy R Reed [EMAIL PROTECTED] wrote: How about setting up a bounty? http://voip-info.org/wiki-Asterisk+bounty If I had the money to rent-a-coder, would I have begged on a public mailing list? You are missing the point. If you contribute $20 and everyone else who wants this feature puts up their $20 it eventually adds up to enough money to get someone interested. You don't have to bear the entire burden yourself. I'm not sure if software support for the line reversal monitoring is desirable in the driver. As the bounty calls for the code to be accepted into CVS, this would be an issue for anyone looking to collect. There were rumours, all those years ago :-), that the new FXO modules would support line reversal detection in hardware. I can't find any technical specifications or other documentation for the new modules so I can't tell if the modules are aware of a world outside of the USA. I hope that they have this capability. I also hope that they have the relevant approvals for connection to the PSTN etc. I was rather hoping that, by now, one of the $15 X100P manufacturers would have released a $16 X100P card, with a slightly upgraded chipset, that was suitable for international operation. Digium seem to have more of a profit margin with their $100 version, so they could probably release an updated card (X102P?) without it affecting the price. At the moment, the choices seem to be: 1. Throw the X101P in the bin and spend $133 on a TDM01B (if it does turn out to support non-USA Caller*ID). 2. Wait for a $16 X100P clone, with an upgraded chipset, and throw the Digium X101P in the bin. 3. Get the soldering iron out and hack serial port support into a BT CD50. 4. Get a digital line (ISDN etc.) Options 1 and 2 still rely upon driver support, but I'd expect the driver changes to be identical, or very similar, no matter which option was taken. ___ 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: Caller ID
On Sunday 02 May 2004 12:30, Kevin Walsh wrote:
> I'm not sure if software support for the line reversal monitoring is
> desirable in the driver.
Probably not - but surely isn't is a well-accepted method that software is
often used to fix bad hardware design?
I firmly believe that half the reason so many companies refuse to open-source
their drivers is because it would horrify us all at the number of "special
case" if {} blocks and /* fix for lack of blablabla support */ comments that
we'd find :)
From what I read, isn't the polarity switch fairly redundant? Would it not be
enough to just leave the fskmodem.c code running all the time (for a single
X100P on a 1GHz system, I can't believe this will have a noticable CPU hit),
instead of the current 'wait for a ring, then listen for CLID' ?
I have tried to look into the source, but I'm no C coder and just ended up
getting increasingly confused :(
Indeed, would there really be much extra work required, since the UK FSK
databurst is in the same format as the USA (from what I understand) - so much
less work would be required to suport this, than the DTMF-based CLID...
Cheers,
Gavin.
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[Asterisk-Users] nat friendly tftp server
Hi, I was searching along the net for a tftp server "nat-friendly", in order to provide new firmware to our budgetones, which are 90% of time nat'ed. I came across this one: http://troja.ath.cx/~zond/jtftp/ works ok. is written is java, under linux. I'm using it now and seems ok. very simple, but does the job. Matteo. -- -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl ___ 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] nat friendly tftp server
I personally download the TFTP server provided free by SolarWinds. I simply open up the required port and is very happy. http://support.solarwinds.net/updates/New-customerFree.cfm Regards, Ian. --- Outgoing mail is scanned but not certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.672 / Virus Database: 434 - Release Date: 28/04/2004 ___ 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] nat friendly tftp server
sure... but the one I suggested is meant to be run as a service on a linux server, maybe the same as your asterisk box. why having a winzoz machine to provide only tftp service ? :) Matteo. Il dom, 2004-05-02 alle 14:05, Ian Pilkington ha scritto: > I personally download the TFTP server provided free by SolarWinds. > > I simply open up the required port and is very happy. > > http://support.solarwinds.net/updates/New-customerFree.cfm > > Regards, > > Ian. > > > --- > Outgoing mail is scanned but not certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.672 / Virus Database: 434 - Release Date: 28/04/2004 > ___ > 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 -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl ___ 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] iconnecthere behind NAT, strange deal
> Is anyone on the list successfully using iconnecthere > behind NAT? Yes (unless it broke during the last 12 hours). I gave my daughter a SIP phone and an Iconnecthere account. I have successfully used Sipura 2000, a Grandstream and a Cisco ATA box with that account, and yes, she is behind a *standard* LinkSys NAT router which itself gets a dynamic WAN address from the service provider. Now, that being said, I am not totally happy with Iconecthere. Their service availability has been spotty, and they can not seem to be able to forward the correct (outgoing) callerID. > I was, for over a year, and then I changed my "plan" with > them. Now all my calls get intercepted immediately, What 'plan' change did you make? > FWIW I used to prepend "" to the dialed number, and it > worked fine until last week. I don't recall having to do that. Iconnecthere just gives us (USA based) international dialtone. Willy - Original Message Follows - > I've been to the WIKI and I've searched the archives. > > Is anyone on the list successfully using iconnecthere > behind NAT? > > I was, for over a year, and then I changed my "plan" with > them. Now all my calls get intercepted immediately, > "We're sorry, but your account is temporarily > unavailable." > > Incoming calls work just fine. > > I contacted their so-called "customer care," which has > sent me repeated replies asking me to give them the > version of my PC phone. When I say I don't have one, > they say, "Sorry, we only help those who do." > > I like to play with their GSM stuff, so I hate to let the > account go, but if no one here knows what might be going > on, they certainly don't. > > FWIW I used to prepend "" to the dialed number, and it > worked fine until last week. > > Thx. > > B. > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Talking SIP to Vocal
I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I have: exten => 150,1,Dial(SIP/[EMAIL PROTECTED]) When I call 150 Asterisk sends an invite to Vocal which then asks for authentication. Asterisk sends another invite with auth details *but* the digest username is "0800800150" when it should (I think) be "08452416761". I'm using source from CVS, checked out yesterday. Calls out via IAX work fine. Calls out via SIP to Free World Dialup work fine, but then FWD doesn't ask for authentication. Is this a bug in the SIP auth code or am I misconfiged? Any ideas please? Thanks, Mark. ___ 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] AGI question
Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ 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] Grab phone call ?
Hi Let's say i have a call to a extension 115. But i'm under the extension 118 how take the call from 115 to my extension using * ?? Thanks alot. ___ 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] Simple SIP X-Lite Configuration Failing
I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening:
localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtestMay 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congesting SIP/jtest-6a1e -- SIP/jtest-6a1e is circuit-busy == Everyone is busy at this timeMay 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No application 'DialCongestion' for extension (sip, 22, 2) == Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
My setup is very simple and basic:
SIP.conf
[general]port = 5060bindaddr = 0.0.0.0context = sip; Default
[jay]type=friendsecret=jaysipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay <400>"disallow=allallow=gsmcontext=sip
[jtest]type=friendsecret=jaytestsipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay Test <410disallow=allallow=gsmcontext=sip
extensions.conf
[sip] ; context for X-Lite Clientsexten =>11,1,Dial(SIP/jay,20,tr)exten =>11,2,Congestionexten =>22,1,Dial(SIP/jtest,20,tr)exten =>22,2,DialCongestion
Lastly, here's my client setup
Display Name: JayUser Name & Authorization User: jay
Password: jaysip Domain/Realm: 192.168.1.20 SIP Proxy: 192.168.1.20
Display Name: Jay TestUser Name & Authorization User: jtest
Password: jaytestsip Domain/Realm: 192.168.1.20 SIP Proxy: 192.168.1.20
Any help anyone can give me would be appreciated since I've already spent HOURS on this and have made absolutely no progress in debugging this (didn't find anything in any of the archives nor wiki pages).
J...
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Re: [Asterisk-Users] Playing with time ranges...
On Saturday 01 May 2004 18:00, Hermann Wecke wrote: > On Fri, 30 Apr 2004, Mark Elkins wrote: > > Looking at pbx.c - I'm not sure if I should change the end time > > (ie midnight) to either 23:59 -or- 00:00. > > it is 23:59 > > > 23:59 will work - but what happens to calls then between 23:59 > > and midnight? > > 23:59'59" is still 23:59 mainly because you are not handling > "seconds", pnlu hours and minutes. > > And, actually, 24:00 does not exist... > > This about this recipe: > > include => night_menu|17:00-23:58|Mon-Thu|*|* > > in this example, you are starting at 17 hours, 0 minutes 0 seconds > and ending at 23 hours, 58 minutes, 59 seconds.. Actually, you aren't. If you take a look at the underlying logic, the minimum increment is 2 minutes, which means that 23:58 is treated the same as 23:59. -- 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
Re: [Asterisk-Users] Simple SIP X-Lite Configuration Failing
J Poz wrote: -- SIP/jtest-6a1e is circuit-busy == Everyone is busy at this time May 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No applica tion 'DialCongestion' for extension (sip, 22, 2) Come on... Asterisk is telling you EXACTLY whats wrong... SIP/jtest is circuit-busy and then asterisk cannot find an application named DialCongetsion. Fix your extensions.conf Jeremy McNamara ___ 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] nat friendly tftp server
On Sun, 2004-05-02 at 13:48 +0200, Brancaleoni Matteo wrote: > Hi, > > I was searching along the net for a tftp server > "nat-friendly", in order to provide new firmware > to our budgetones, which are 90% of time nat'ed. > > I came across this one: > http://troja.ath.cx/~zond/jtftp/ > I found gs_config on the same site more interesting it's working with my standard tftp. Although I can't at the moment work out all the changes necessary for 1.0.4.63. The other problem is how to write to different phones by MAC using cfg.txt -- Dave Cotton <[EMAIL PROTECTED]> ___ 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] Simple SIP X-Lite Configuration Failing
Jeremy, I'm missing something hereI understand the congestion part - no problem, I can take that out of extensions.conf.. But I need help in the real problem which is the Auto-congesting and circuit-busy part. Why is it saying the circuit is busy? If siphone "jay" dials "jaytest" it says circuit-busy eventhough jaytest is not busy (and vice-versa). So what am I missing (I know this is simple but I haven't been able to figure it out so why I'm asking for help). J..Jeremy McNamara <[EMAIL PROTECTED]> wrote: J Poz wrote:> -- SIP/jtest-6a1e is circuit-busy> == Everyone is busy at this time> May 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: > No applica> tion 'DialCongestion' for extension (sip, 22, 2)Come on... Asterisk is telling you EXACTLY whats wrong... SIP/jtest is circuit-busy and then asterisk cannot find an application named DialCongetsion.Fix your extensions.confJeremy McNamara___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
Re: [Asterisk-Users] Talking SIP to Vocal
Hi! > I'm trying to get Asterisk to talk SIP to Vocal and so far have only > managed to get it partially working. Calls in from Vocal are working > fine but outbound calls aren't. I haven't looked at your settings, but two days ago I upgraded to latest CVS and since then I am unable to place outgoing calls with Nikotel - the rest of the SIP calls seem to work fine, however, including FWD. Hopefully tomorrow I'll be able to dig a bit deeper into this. Cheers, Philipp ___ 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] Fax Detect problem (have consulted archives, wiki & irc)
On 1-May-04, at 4:51 PM, Jeb Campbell wrote: On May 1, 2004, at 2:04 PM, Ryan Courtnage wrote: I'm trying to get fax detection to work. Hi Ryan, in stable FAX_DETECT is turned off by default in the code (dsp.c). I'm personally using it with spandsp and having no problems, but YMMV. If you want to enable it, goto line 60 of dsp.c and uncomment that #define so it looks like this: /* Define if you want the fax detector -- NOT RECOMMENDED IN -STABLE */ #define FAX_DETECT Then recompile and install. Jeb, thanks for taking the time to reply - i could have easily killed days looking for this! It's working fine now, but indeed, my mileage may vary ... which brings up the question; What is the reason FAX_DETECT is not recommended in Stable? Cheers, Ryan Jeb Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ 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] Talking SIP to Vocal
Philipp von Klitzing wrote: Hi! I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. I haven't looked at your settings, but two days ago I upgraded to latest CVS and since then I am unable to place outgoing calls with Nikotel - the rest of the SIP calls seem to work fine, however, including FWD. Hopefully tomorrow I'll be able to dig a bit deeper into this. I think the username/secret items in sip.conf are busted. A quick ethereal trace shows that when placing an outbound call to another provider via SIP, * is not using the username defined during the authentication challenge, instead it uses the username of the phone placing the call. A rollback to CVS of a week ago fixes the issue. Cheers, Philipp ___ 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 -- Andres Network Admin http://www.telesip.net ___ 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] New ENUM service, what do you think?
On Sat, 1 May 2004, Dean Collins wrote: > Yes but no information about how this will operate, what regulation or > restrictions on joining, what connection protocols will be used etc etc agreed - you see alot of business fluff - but the technicals are very important and on many of these ventures they fail to include them. regards joe www.baptista.god > > Cheers, > Dean > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Reid A. > Forrest > Sent: Saturday, 1 May 2004 8:21 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] New ENUM service, what do you think? > > >From http://www.thevpf.com/ > > To join, please e-mail [EMAIL PROTECTED] or telephone 1-212-232-2020 > (Mon-Fri > 9AM-5PM EST). > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of jimfl > Sent: Saturday, May 01, 2004 5:11 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] New ENUM service, what do you think? > > >Jim/frank, > >Can you give us more information about how to access this enum? I've > >been to the stealth web site and there is no information about access. > > > >I look forward with interest to what you have up and running today for > >asterisk users to benefit from. > > > >Cheers, > >Dean > > Sorry, I am not associated with Stealth in any way. Just saw the news > story > and > thought it would be of interest to Asterisk users. It sounds like you > don't > have to > be a VOIP provider to get access to their service. They talk about > businesses > using the service. If anyone finds out how to get access to their > service, > please > post. > > Jim > > ___ > 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] Using IAXTel to dial FWD
On Sat, 1 May 2004, Rich Adamson wrote: > > > now. But if you have a look at this page -> > > http://www.freeworlddialup.com/advanced/iax you will find that you can now > > use FWD with IAX2 along with SIP :) > > FWIW, I just moved our FWD account to iax2, and it works rather well > with *. The referenced web page does have a couple of configuration > errors on it, but nothing all that difficult to diagnose/fix. > > Also, it appears the FWD -> IaxTel definitions are incorrect (again) > causing problems with connectivity in both directions. Could you share your conf files with us. I can connect to FWD on my asterisk - but FWD only see me as an external SIP agent and not a SIP client of the FWD network. DOn't know exactly why - so would luv to compare your conf files. thanks joe Joe Baptista: USG Portal www.joebaptista.com, Personal www.baptista.god LOW: low cost, Low Lands. http://www.dot.low/ Everything you need to know about LOW ___ 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] Fax Detect problem (have consulted archives, wiki & irc)
On May 2, 2004, at 1:02 PM, Ryan Courtnage wrote: I'm personally using it with spandsp and having no problems, but YMMV. If you want to enable it, goto line 60 of dsp.c and uncomment that #define so it looks like this: /* Define if you want the fax detector -- NOT RECOMMENDED IN -STABLE */ #define FAX_DETECT Then recompile and install. Jeb, thanks for taking the time to reply - i could have easily killed days looking for this! It's working fine now, but indeed, my mileage may vary ... which brings up the question; What is the reason FAX_DETECT is not recommended in Stable? No idea? The number of people using it seems pretty small (compared to all * use), so if they were having problems, if would make sense to change the default. But really I have no idea -- and it is working great for me! Jeb ___ 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: Caller ID
Steve Underwood [EMAIL PROTECTED] wrote: > Wake up. > Sorry, I must have drifted off for a while. Thanks for the alarm call. > > The reversal detection is a complete waste of time. Totally unnecessary. > Pointless. A line break detector would have much more use, as it would > give a reliable disconnect detection on many lines. (Actually, reversal > detection would have years ago, but its not much use any more). > Perhaps both would be good then: A polarity reversal detector for determining the start of a Caller*ID sequence and a line break detector for, err, detecting line breaks. Actually, my X101P seems to detect hangups just fine, so I've not had cause to check whether the detection is done in the hardware or in the driver. If you say that it's not done in hardware then I'll take your word for it for the moment. > > All you need for these CLI requirements is to monitor for some energy on > the line. Since these FXOs are not being used in banks of hundreds, you > will never notice this MIPs this uses. > I'd still prefer to see this done in hardware, rather than in some sort of idle loop in the driver or the application. Call me old fashioned, but I prefer it when unnecessary overheads are not measured in MIPs. :-) Perhaps the new FXO module for the TDMxxB has, or will have, hardware support for the above. If it has, and I'm sure I heard somewhere that it does, then that's great. An X102P, with similar support, would no doubt be welcome too. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] phonejack and linejack in the same system
If you are indeed running RH9.0 you don't need to install any extra ixj drivers. This distro has all drivers included and they work fine. ztdummy has nothing to do with chan_phone. However, you do need to create the device nodes and load the driver with modprobe. It wasn't clear to me from the post that you did that. to make sure you actually have a working device, try this before running asterisk: as root run: cat /proc/ixj It should list all the device parameters. If you are running more than one card, you have to be careful of the limitations of the config file, the parameter apply to all cards. Make sure the device= lines are at the end of the file. I have a rewritten driver for chan_phone, that corrects all these problems and lets you configure each device separately. You are welcome to try it. Send me a private e-mail if you're interested. Karl Brose - Original Message - From: "Raul Elizondo (wizardteam)" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, May 02, 2004 06:03 Subject: RE: [Asterisk-Users] phonejack and linejack in the same system > This is as far as i could go... > > I readed about the archives for "ztdummy", and made the fix and recompiled > after using ./configure --disable-isa-pnp, i edited the phone.conf and > extensions.conf, but i still get this messages: > > > [chan_phone.so] => (Linux Telephony API Support) > == Parsing '/etc/asterisk/phone.conf': Found > May 2 03:52:44 WARNING[16384]: chan_phone.c:950 mkif: Unable to open > '/dev/phone0' > May 2 03:52:44 ERROR[16384]: chan_phone.c:1141 load_module: Unable to > register channel '/dev/phone0' > == Unregistered channel type 'Phone' > May 2 03:52:44 WARNING[16384]: loader.c:326 ast_load_resource: > chan_phone.so: load_module failed, returning -1 > == Unregistered channel type 'Phone' > May 2 03:52:44 WARNING[16384]: loader.c:421 load_modules: Loading module > chan_phone.so failed! > [EMAIL PROTECTED] asterisk]# ls /dev/phone0 > /dev/phone0 > > I am going to compile openh323 just to make sure both quicknet cards work in > this box, but it was working in another box with redhat9. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Raul Elizondo > (wizardteam) > Sent: Sunday, May 02, 2004 3:28 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] phonejack and linejack in the same system > > > i actually found a couple things interesting in the archives after i wrote > my first email, and i deleted the soundcard, but i still got problems with > the linejack, i m testing all possible options. > > -=Raul=- > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk > Sent: Sunday, May 02, 2004 1:45 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] phonejack and linejack in the same system > > > I'm a newbie too -- search the archives for "ztdummy". > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Raul > Elizondo (wizardteam) > Sent: Sunday, May 02, 2004 2:32 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] phonejack and linejack in the same system > > > Hi, > > I am a newbie in asterisk, i could compile it and run it with no problem > on a RedHat 9. In the same box, i got a linejack and a phonejack cards > and i downloaded the CVS driver from quicknet. This 2 card were working > in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. > > I am using the default samples, and i tried /dev/phone0 and /dev/phone1, > but when i run asterisk, i get this error: > > [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) > May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to > open IAX timing interface: No such device > == Manager registered action IAXpeers > == Parsing '/etc/asterisk/iax.conf': Found > == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver > 2)) > == Using TOS bits 16 > == IAX Ready and Listening on 0.0.0.0 port 4569 > == Loaded firmware 'iaxy.bin' > [chan_local.so] => (Local Proxy Channel) > == Registered channel type 'Local' (Local Proxy Channel Driver) > [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) > == Parsing '/etc/asterisk/skinny.conf': Found > == Skinny listening on 0.0.0.0:2000 > == Registered channel type 'Skinny' (Skinny Client Control Protocol > (Skinny)) > [chan_oss.so] => (OSS Console Channel Driver) > May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to > open /dev/dsp: No such device > == No sound card detected -- console channel will be unavailable > == Turn off OSS support by adding 'noload=chan_oss.so' in > /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API > Support) > == Parsing '/etc/asterisk/phone.conf': Found > May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open > '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 > load_module: Unable to register channel '/dev/ph
Re: [Asterisk-Users] Using IAXTel to dial FWD
Joe, Inline comments... > > > now. But if you have a look at this page -> > > > http://www.freeworlddialup.com/advanced/iax you will find that you can now > > > use FWD with IAX2 along with SIP :) > > > > FWIW, I just moved our FWD account to iax2, and it works rather well > > with *. The referenced web page does have a couple of configuration > > errors on it, but nothing all that difficult to diagnose/fix. > > > > Also, it appears the FWD -> IaxTel definitions are incorrect (again) > > causing problems with connectivity in both directions. > > Could you share your conf files with us. I followed the instructions on the fwd page, but I had to add "allow=ulaw" to the iax.conf general section (which was hinted on the page but easily missed), and changed the Dial statements (since only one of them is needed). I also wrote back to FWD regarding this and Ed has updated those fwd pages as of yesterday. Might compare your config files to his again. What he is showing on the pages right now should work just fine; they do work for me. > I can connect to FWD on my asterisk - but FWD only see me as an external > SIP agent and not a SIP client of the FWD network. DOn't know exactly why > - so would luv to compare your conf files. I don't know that I'm understanding your statement above relative to sip. Taking a guess... one of the steps on the page indicates you have to write an email to Ed and ask him to make the changes so your account uses iax2 instead of sip. That note suggests it might take a day or two for those manual changes to be completed. It could be your changes have not been completed as yet, or you didn't send him the email; tough to guess based only on the words noted above. If you previously had a register statement in your sip.conf for fwd, you should comment it out as its no longer needed after the iax conversion. The following steps should help you diagnose what is happening: 1. I don't remember which asteisk config file changes get reread when you do an asterisk "relaod" command from the CLI, so just get in the habbit of "stop now" and restart asterisk from scratch when making the changes needed for this sip to iax conversion. 2. If your register statement in iax.conf is correct, then when you execute the "iax2 show registry" it should return something like: Host UsernamePerceived Refresh State 65.39.205.121:456962353 203.212.183.101:4569 60 Registered where 62353 represents "your" fwd number (whatever it is). If you don't see the above, then fix your register statement and stop/restart asterisk. 3. FWD site indicates that dialing 612 will provide you with the current time. Assuming you followed the instructions correctly on their page, dial that number from one of your phones, then execute "iax2 show channels" while the call is in progress. You should see something like: Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter Format 65.39.205.12162353 1/1 3/7 0ms 0019ms ULAW 1 active IAX channel(s) The above would confirm your "outbound" entries in extensions.conf and iax.conf are correct. If you don't see the above, then go back and check your configs again. 4. The FWD site includes a page that sends a test call to your asterisk box. Find it and use it to generate an incoming call, which will test all remaining configuration statements on your system. Its currently about half the way down the page at: http://www.freeworlddialup.com/content/view/full/274/ where it says "Call Me". It is a handy way to make inbound call tests without bothering other FWD users. 4. Last, you should be able to originate a call to your own FWD number and watch the asterisk CLI send that call to FWD, and FWD attempting to call your phone. Read those statements very carefully as the words will actually describe most configuration problems. If you want to test further, then initiate a FWD call from some other phone other then the one that you normally receive incoming FWD calls from. You should be able to call yourself via FWD nicely. If you still can't get this to work, then give us some real output from your asterisk CLI, and paste portions of the extensions.conf and iax.conf that relate to this. Rich ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Saturday 01 May 2004 16:42, Gavin Hamill wrote: > > PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE can some Kind and Worthy > soul spend a little time in getting this really important feature > implemented? You would have the undying gratitude of thousands of X100P > users all round the world! :D I emailed sales at digium asking whether the new module supported international (ie non bellcore) cli. The answer was yes, but it's not yet implemented in the driver - driver implementation is in the pipeline apparently. Whether this means that the detection is in the hardware or not I don't know. Jon ___ 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: Caller ID
I've come late into this thread, so I risk saying things that you all will just shake your heads at and say, "Duh!" Historically, though, from WAY back in the days of electromechanical switches, reverse battery mainly provided answer supervision. Its usefulness pretty much went away with the advent of SS7, except for those cases where end users resold their POTS services (such as hotels and motels, which usually paid extra for the service). The battery would then reverse BACK to normal again after the call was terminated. During this reversal (obviously), the voltage would transition past zero, and it would also suffice for disconnect supervision. Aside from the hotel/motel scenario, telcos have recently been providing disconnect supervision solely by means of removal of battery from the circuit. This feature continues to be of value in situations where analog CPE would continue to keep the line seized were it not for the removal of the battery -- key systems having lines on hold, answering machines, etc. Personally, I would very much like to see the reverse battery feature built in to the FXS cards that work on asterisk. I say this because I am starting to go back to my roots in the industry by looking for old step-by-step line finders, selectors, and connectors. Answer supervision via some electromechanical means would be preferable than trying to cobble an ISDN D channel over to that old stuff. Just my 2 cents. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Kevin Walsh > Sent: Sunday, May 02, 2004 1:05 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] RE: Caller ID > > Steve Underwood [EMAIL PROTECTED] wrote: > > Wake up. > > > Sorry, I must have drifted off for a while. Thanks for the > alarm call. > > > > > The reversal detection is a complete waste of time. Totally > unnecessary. > > Pointless. A line break detector would have much more use, > as it would > > give a reliable disconnect detection on many lines. (Actually, > > reversal detection would have years ago, but its not much > use any more). > > > Perhaps both would be good then: A polarity reversal > detector for determining the start of a Caller*ID sequence > and a line break detector for, err, detecting line breaks. > Actually, my X101P seems to detect hangups just fine, so I've > not had cause to check whether the detection is done in the > hardware or in the driver. If you say that it's not done in > hardware then I'll take your word for it for the moment. > > > > > All you need for these CLI requirements is to monitor for > some energy > > on the line. Since these FXOs are not being used in banks > of hundreds, > > you will never notice this MIPs this uses. > > > I'd still prefer to see this done in hardware, rather than in > some sort of idle loop in the driver or the application. > Call me old fashioned, but I prefer it when unnecessary > overheads are not measured in MIPs. :-) > > Perhaps the new FXO module for the TDMxxB has, or will have, > hardware support for the above. If it has, and I'm sure I > heard somewhere that it does, then that's great. An X102P, > with similar support, would no doubt be welcome too. > > -- >_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ > _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n > W a l s h > _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] > _/ _/ _/_/_/_/ _/_/_/_/ _/_/ > > ___ > 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] chan_vpd patches
Who ever works with chan_vpd please look over this: http://bugs.digium.com/bug_view_page.php?bug_id=961 Thanks,Brian
Re: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Sunday 02 May 2004 19:33, Jon Lawrence wrote: > I emailed sales at digium asking whether the new module supported > international (ie non bellcore) cli. The answer was yes, but it's not yet > implemented in the driver - driver implementation is in the pipeline > apparently. > Whether this means that the detection is in the hardware or not I don't > know. Uh huh - and as for those who have already 'Supported Digium' by buying their hardware, will they get a free upgrade, or will we all be expected to 'Support Digium' again? I feel the least they can do is devote a few of those "hours of support" the X100P buys into adding CLID support for the UK 'nearly-Bellcore' variant, and the DTMF-based ones. So it needs to leave a 1200baud FSK modem running all the time - so what? It's a software modem anyway, what noticable difference is another few hundred lines of software going to make? This is a technical forum, but it's increasingly difficult to resist the temptation to bring the "$world = $usa" aspect in :( Cheers, Gavin. ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
WARNING!!! WARNING!!! WARNING!!! If support for UK callerid is a MUST then get off your butt and raise some hell. I sit here and read thru these emails where people chatter back and forth about it. I don't see anyone trying to drum up intrest in this! If this is something you want MAKE IT HAPPEN. I'm not saying code it, I'm not saying PAY for it. I'm saying find someone that can make it happen and ask for help. I didnt' know a line of C when I started with asterisk but now I can do the most basic things and even fix stuff. (I even wrote cdr_odbc.c which btw was my first C project in asterisk and is in CVS) So basically what i'm saying is PUT UP or SHUT UP. Don't stand by bitching about something you wanna see, jump in head first and try your best to make it happen. Thats what I do. bkw Asterisk Bug Marshal #asterisk-bugs ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Sunday 02 May 2004 21:16, brian k. west wrote: > I'm not saying code it, I'm not saying PAY for it. I'm saying find someone > that can make it happen and ask for help. Believe me I'm already doing this - I've rallied the local Linux geek community as much as I can and there are a couple of interested parties whom I'm still pursuing :) I'm rattling on here because it's the single biggest focus point for people interested in Asterisk, and I'd hoped by merely continuing the thread, that someone might just be motivated enough to crack open a copy of vi and toy with it for a while. > So basically what i'm saying is PUT UP or SHUT UP. Don't stand by bitching > about something you wanna see, jump in head first and try your best to make > it happen. Thats what I do. I agree it's painful to watch threads like this (especially so when you're the source of them - this type of thread isn't my usual style), but I *am* doing what I can, and from my own history in support, I've found that if a customer/client/whoever cries long enough and loud enough, then either someone will take pity on them, or fix the 'damn bug' just to shut them up. Perhaps I've just misread the * development base - I'm sorry if that's the case. As for C code - I've worked through the codebase to the best of my ability - even if someone could say "start on line n of file x.c" I might be in a better position to attack it. Right now all I'd like to do is move the fskmodem.c code from "start after first ring" to running all the time. I feel inadequate enough as it is that I'm unable to do this - a little push in the right direction could go a long way. Cheers, Gavin. ___ 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] Using IAXTel to dial FWD
> I can connect to FWD on my asterisk - but FWD only see me as an external
> SIP agent and not a SIP client of the FWD network. DOn't know exactly why
> - so would luv to compare your conf files.
>
There is really not much you can do about that with your local
configuration.
The problem is on FWD's Asterisk, it doesn't properly log an IAX user into
the
SIP Express router. The only service that is broken because of it seems to
be
PSTN access. There has been a long discussion about this in the FWD forum.
You can see this in your received calls log on FWD. The registration realm
is
not proper, it's just the IP address of the asterisk server.
The following configuration should help those who would like to give this a
try.
Finally, a note regarding the FWD setup with unauthenticated calls and host
access
only, as shown below in the [fwd-in] section and the FWD website:
Technically this is an incorrect configuration and security hole, as there
should be some
authentication token, either a secret or a key (as the IAXTEL people are
using)
Since FWD IAX was announced, the chan_iax driver has been changed to allow
this without failure if you have more than one such client in your config
file (it had to
be the last one, before the change, or you had to accept the call totally
unauthenticated
through the default context, which was a major bug.
Note also that you have to have g711 enabled with FWD.
IAX.CONF
==
[general]
; .. other things here most likely
;
disallow=all
allow=ulaw; and perhaps others
register=##FWDNO##:[EMAIL PROTECTED]
[fwd-in]
type=user
deny=0.0.0.0/0.0.0.0
permit=65.39.205.0/255.255.255.0
accountcode=IAXFWD.01
context=fwd-iax-inbound
[fwd-peer]
type=peer
host=iax2.fwdnet.net
username=##FWDNO##
secret=**
accountcode=IAXFWD.02
qualify=yes
context=NOACCESS
EXTENSIONS.CONF
==
[some_where_in_your_dial_plan]
exten=_**393.,1,Macro,fwd_iax_call,${EXTEN:5}
[fwd-iax-inbound]
exten=##FWDNO##,1,Answer
;and whatever you want to do for an incoming call
[macro-fwd_iax_call]Macro to place an IAX call through FWD
; ${ARG1} - number to call
exten= s,1,SetCIDNum(##FWDNO##)
exten= s,2,SetCIDName(##your name##)
exten= s,3,Dial(IAX2/[EMAIL PROTECTED]/${ARG1})
exten= s,4,Hangup
exten= s,104,Hangup
exten= s,105,Congestion
___
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[Asterisk-Users] Cisco 12SP+
Hi, I'm thinking about getting a couple of Cisco 12SP+ phones to use on my Asterisk system. I have just bought a Cisco 7960, and they are great, but too expensive to buy a lot of them, so I though I might try the 12SP+ ones. I have seen in the archives that the phones work on Asterisk, but I can't see much in there about the images in use. When I got my 7960, it had the call manager image on it, and I had to convert it to the SIP image before I could use it. Is this the same case with the 12SP+, do you need to change it's image ? Thanks in advance, Paul.
RE: Caller ID Re: [Asterisk-Users] Re: Support Digium
Jon Lawrence [EMAIL PROTECTED] wrote: > I emailed sales at digium asking whether the new module supported > international (ie non bellcore) cli. The answer was yes, but it's not yet > implemented in the driver - driver implementation is in the pipeline > apparently. Whether this means that the detection is in the hardware or > not I don't know. > I'll take that to mean that the support is in the hardware, which is very good news. The Digium shop (http://store.yahoo.com/asteriskpbx/newitd1pofxo.html) says that I must have PCI 2.2 to make use of the card, so that probably rules out the current machine I'm using, which is an old Pentium II 300-ish. I don't really know how to tell which PCI version that server is using, but I doubt that it's up to date. Someone on IRC once pointed out the conflict between "suggested" and "must" on a similar page and said that their TDM400P (FXS-only at the time) was working on a PCI 2.1 system. Can anyone confirm whether a PCI 2.2 bus is mandatory? Perhaps I need to throw the server out along with the X101P card before I can order a TDM400P and a shiny new FXO module. The same Digium shop page suggests that two PCI slots would be required. I'll assume the card is too fat, with the daughter board(s) fitted, to fit into a single slot. The "BT CD50 and soldering iron" plan is looking more and more like the one I'll be going with for now. If anyone has a pre-modified (and working) BT CD50, or similar, and would like to sell it to me then please feel free to email me off-list, otherwise I'll order the parts and have a go myself. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] FXO line hum w/ Z-plex 10
I've recently begun integrating an Asterisk system into my house. I purchased the Dev Kit a year or two ago when Digium was selling it as a Z-plex 10 channel bank with the T100P. I've recently found that when I connect the serial monitoring port to my system it introduces a noticable hum on my incoming FXO lines. Anyone know (or have suggestions about) how to prevent this? -- Jamin W. Collins To be nobody but yourself when the whole world is trying it's best night and day to make you everybody else is to fight the hardest battle any human being will fight. -- E.E. Cummings ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
brian k. west [EMAIL PROTECTED] wrote: > If support for UK callerid is a MUST then get off your butt and raise some > hell. I sit here and read thru these emails where people chatter back and > forth about it. I don't see anyone trying to drum up intrest in this! If > this is something you want MAKE IT HAPPEN. > > I'm not saying code it, I'm not saying PAY for it. I'm saying find > someone that can make it happen and ask for help. I didnt' know a line > of C when I started with asterisk but now I can do the most basic things > and even fix stuff. (I even wrote cdr_odbc.c which btw was my first C > project in asterisk and is in CVS) > > So basically what i'm saying is PUT UP or SHUT UP. Don't stand by > bitching about something you wanna see, jump in head first and try your > best to make it happen. Thats what I do. > Good for you. I know 'C', but hardware support is a prerequisite here. It seems that the new FXO modules have the required hardware support so, at first glance, that appears to be a step in the right direction. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Sun, 2004-05-02 at 22:07, Kevin Walsh wrote: > Jon Lawrence [EMAIL PROTECTED] wrote: > > I emailed sales at digium asking whether the new module supported > > international (ie non bellcore) cli. The answer was yes, ... > The Digium shop (http://store.yahoo.com/asteriskpbx/newitd1pofxo.html) > says that I must have PCI 2.2 to make use of the card,... > The "BT CD50 and soldering iron" plan is looking more and more like the > one I'll be going with for now Um - Digium wants you to buy their hardware - but there is a CLID issue.. would it not make more financial sense to insert a dumb ISDN card (or two), and upgrade your PSTN to ISDN??? Would this not "assist" Digium in making sure CLID worked in the UK??? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] IAX2
What does it mean ? May 2 20:37:21 WARNING[1205250992]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386, dropping Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux from cvs checkout -r v1-0_stable asterisk
Re: [Asterisk-Users] IAX2
I think this was fixed in CVS-HEAD because I do not see that message in the src at all while looking to see if t was fixed. bkw - Original Message - From: Serge Oleinikov To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 2:40 PM Subject: [Asterisk-Users] IAX2 What does it mean ? May 2 20:37:21 WARNING[1205250992]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386, dropping Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux from cvs checkout -r v1-0_stable asterisk
[Asterisk-Users] TDM400P FXO, 2 slots?
On Sun, May 02, 2004 at 09:07:37PM +0100, Kevin Walsh wrote: > > The same Digium shop page suggests that two PCI slots would be required. > I'll assume the card is too fat, with the daughter board(s) fitted, to > fit into a single slot. This is something I would like to see confirmed, does this card really take 2 pci slots? I had hoped to make use of one of these and a T100P in an SS40G case for personal home use. -- Jamin W. Collins "Never underestimate the power of very stupid people in large groups." -- John Kenneth Galbraith ___ 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] TDM400P FXO, 2 slots?
> > The same Digium shop page suggests that two PCI slots would be required. > > I'll assume the card is too fat, with the daughter board(s) fitted, to > > fit into a single slot. > > This is something I would like to see confirmed, does this card really > take 2 pci slots? I had hoped to make use of one of these and a T100P > in an SS40G case for personal home use. I just installed the new card the other day, and it only uses a single slot. Via visual inspection (just now) it would appear to fit in any pci slot, however with the daughter boards on it, it would appear that a sheet of cardboard or something might help keep it from bumping the next pci card. Its pretty fat, but doesn't lap into the space of the next slot. ___ 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] problems with bri-stuff.0.0.2rc20a
hi everybody, just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange problem :-( i have immediate = no but when i pickup the phone i get : *CLI> == D-Channel on span 1 up -- Extension 's' in context 'default' from '6294094' does not exist. Rejecting call on channel 2, span 1 i have started asterisk with -vvc so there should be a debug message if immediate mode was on. maybe anyone (klaus-peter) can help. i'm using a hfc-card in nt-mode. i'm not 100% shure but i think that my phone is using uk-tones (ring ...) since the update but all language-settings are nl. looking forward to get some help ;) thorsten ___ 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] module help?
Need some help with modules.conf, and basic RH9 linux skills. I've installed the new TDM04B 4-port FXO card and its working. After a reboot, when I do lsmod I see the wcfxo module but not the wcfxs even though both are listed modules.conf. If I "modprobe wcfxs", then lsmod has both modules showing. The wcfxs module is the last one in the modules.conf. Is the order of entries sensitive in modules.conf? Do I need to be concerned with wcfxs not showing before starting asterisk? Any suggestions? Rich ___ 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] module help?
Hi. > Need some help with modules.conf, and basic RH9 linux skills. perhaps wrong list? see linux kernel howto... > I've installed the new TDM04B 4-port FXO card and its working. After > a reboot, when I do lsmod I see the wcfxo module but not the wcfxs > even though both are listed modules.conf. > > If I "modprobe wcfxs", then lsmod has both modules showing. why you need wcfxs on a quad-fxo ? > The wcfxs module is the last one in the modules.conf. Is the order > of entries sensitive in modules.conf? modules.conf != loaded modules. as the name suggest, it contains only configuration params for modules > > Do I need to be concerned with wcfxs not showing before starting > asterisk? Any suggestions? sure. learn something more about kernel, modules and what is modules.conf bug us with asterisk related questions, not with what-are-kernel-modules? questions. -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
Mark J Elkins wrote >Um - Digium wants you to buy their hardware - but there is a CLID >issue.. would it not make more financial sense to insert a dumb ISDN >card (or two), and upgrade your PSTN to ISDN??? Would this not "assist" >Digium in making sure CLID worked in the UK??? Isn't this a bit like cutting of the nose to spite the face. UK PSTN lines costs £30 /Qtr UK ISDN costs £65 /qtr, you could buy two X100P's every year and still be in pocket by staying with PSTN. There was a post on the list in the not to distant past where someone had written two small scripts for getting the information from a BT50 and a serial modification and passing it to asterisk. Still seems the best way in the interim. As has been said many times in the list Digium have given us this software, we don't have to give them a hard time in return. Not a fair payback. Dave ___ 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] Voicemail or voicemail2?
I'm using the stable branch. Is voicemail or voicemail2 deprecated? TKS Paul [EMAIL PROTECTED]
Re: [Asterisk-Users] Voicemail or voicemail2?
Scrive Paul Mahler <[EMAIL PROTECTED]>: > I'm using the stable branch. Is voicemail or voicemail2 deprecated? > RGH!!! ages passed when voicemail was sent to /dev/null and voicemail2 moved to voicemail... current voicemail is the old voicemail2 voicemail doesn't exist any more. perhaps voicemail2 exists only as an alias to voicemail to make the transition smoother Matteo. -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ 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] module help?
> > I've installed the new TDM04B 4-port FXO card and its working. After > > a reboot, when I do lsmod I see the wcfxo module but not the wcfxs > > even though both are listed modules.conf. > > > > If I "modprobe wcfxs", then lsmod has both modules showing. > > why you need wcfxs on a quad-fxo ? Because the support folks at digium said on Friday the supporting routines for the new fxo card are actually in wcfxs. > > The wcfxs module is the last one in the modules.conf. Is the order > > of entries sensitive in modules.conf? > > modules.conf != loaded modules. > as the name suggest, it contains only configuration params > for modules > > > > Do I need to be concerned with wcfxs not showing before starting > > asterisk? Any suggestions? > > sure. > learn something more about kernel, modules and what > is modules.conf > > bug us with asterisk related questions, not > with what-are-kernel-modules? questions. Okay, then let me reword this just for you. Is there a problem with the asterisk "make install" process that might be considered the root-cause for wcfxs not showing up in lsmod? ___ 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] problems with bri-stuff.0.0.2rc20a
immediate=no is in the right position into zapata.conf? ie before the channel=XX you're picking up? matteo. Scrive FastJack <[EMAIL PROTECTED]>: > hi everybody, > > just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange > problem :-( > > i have immediate = no but when i pickup the phone i get : > > *CLI> == D-Channel on span 1 up > -- Extension 's' in context 'default' from '6294094' does not exist. > Rejecting call on channel 2, span 1 > > i have started asterisk with -vvc so there should be a debug message if > immediate mode was on. > > maybe anyone (klaus-peter) can help. i'm using a hfc-card in nt-mode. > > i'm not 100% shure but i think that my phone is using uk-tones (ring ...) > since the update but all language-settings are nl. > > looking forward to get some help ;) > > thorsten > > ___ > 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 > -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Mon, 2004-05-03 at 00:11, David J Carter wrote: > Mark J Elkins wrote > > >Um - Digium wants you to buy their hardware - but there is a CLID > >issue.. would it not make more financial sense to insert a dumb ISDN > >card (or two), and upgrade your PSTN to ISDN??? Would this not "assist" > >Digium in making sure CLID worked in the UK??? > > Isn't this a bit like cutting of the nose to spite the face. > > UK PSTN lines costs £30 /Qtr UK ISDN costs £65 /qtr, you could buy two > X100P's every year and still be in pocket by staying with PSTN. ISDN BRI is two lines - so that makes it £2.50 more per line - or £10 a year..?? no need to purchase the BT50 (a caller-ID unit? - at what cost? you need one per line? and an RS232 interface per unit?) > There was a post on the list in the not to distant past where someone had > written two small scripts for getting the information from a BT50 and a > serial modification and passing it to asterisk. > > Still seems the best way in the interim. > > As has been said many times in the list Digium have given us this software, > we don't have to give them a hard time in return. Not a fair payback. True - the software is excellent. If they sold an ISDN BRI 4-port card (like Fritz) - I'd buy it from them. No intentions of bad mouthing Digium... but USA != World -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a
Hola, if you have overlapdial=no in zapata.conf then * will jump into the s extension on a NT span (this way you can use DigitTimeOut and ResponseTimeOut to make patterns like "_X." work as expected.). So, either you create an s extension, e.g.: exten => s,1,DigitTimeOut(3) or you set overlapdial=yes in zapata.conf. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am So, 2004-05-02 um 23.36 schrieb FastJack: > hi everybody, > > just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange > problem :-( > > i have immediate = no but when i pickup the phone i get : > > *CLI> == D-Channel on span 1 up > -- Extension 's' in context 'default' from '6294094' does not exist. > Rejecting call on channel 2, span 1 > > i have started asterisk with -vvc so there should be a debug message if > immediate mode was on. > > maybe anyone (klaus-peter) can help. i'm using a hfc-card in nt-mode. > > i'm not 100% shure but i think that my phone is using uk-tones (ring ...) > since the update but all language-settings are nl. > > looking forward to get some help ;) > > thorsten > > ___ > 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] Why don't I get a ringing sound?
I am using the following macro to dial a ZAP channel. When I dial in, *
answers and I go to voicemail. I never hear any ringing, though. It doesn't
work with the Ringing command before or after the Dial command.
[macro-zapdial]
;
; call a ZAP extension for ${ARG2} seconds, and then voice mail
; ${ARG1} - Extension
; ${ARG2} - Time to ring
exten => s,1,Dial(ZAP/${ARG1},${ARG2})
exten => s,2,ringing
exten => s,3,Voicemail(u$[${ARG1} + 99]) ; match the channel to the mailbox
exten => s,4,Goto(${ARG1},1) ; If they press #, return to start
exten => s,104,Voicemail(b$[${ARG1} + 99])
exten => s,5,Goto(${ARG1},1) ; If they press #, return to start
Here is what the log shows:
-- Zap/1-1 is ringing
-- Nobody picked up in 2 ms
-- Hungup 'Zap/1-1'
-- Executing Ringing("Zap/49-1", "") in new stack
-- Executing VoiceMail("Zap/49-1", "u100") in new stack
-- Playing 'vm-theperson' (language 'en')
May 2 18:36:45 WARNING[163851]: chan_zap.c:1193 zt_set_hook: zt hook
failed: Device or resource busy
-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (macro-zapdial, s, 3) exited non-zero on 'Zap/49-1' in
macro 'zapdial'
== Spawn extension (main, 100, 1) exited non-zero on 'Zap/49-1'
-- Hungup 'Zap/49-1'
Paul Mahler
[EMAIL PROTECTED]
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RE: [Asterisk-Users] Grab phone call ?
(We prefer plain text posting over HTML posting) ;-) > Let's say i have a call to a extension 115. But i'm > under the extension 118 how take the call from 115 > to my extension using * ?? This has been covered a number of times on the list and in the wiki but it's such a nice day outside I can't be arsed getting all flamey with it.. Short answer: *8 Long answer: It depends on the channel type. Zap channels have a bunch of feature codes hard coded and *8 (pickup) is one of those. You have to have a pickup group defined for each extension in your config file for this to work. For SIP channels, I'm not totally sure - there's been talk of it working, it not working, it working if you patch things.. Check the archives, search for "call pickup" (or "pick up" maybe) and check the wiki at www.voip-info.org/wiki-Asterisk Cheers Paul ___ 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: Caller ID
> The reversal detection is a complete waste of time. Totally > unnecessary. Pointless. A line break detector would have much > more use, as it would give a reliable disconnect detection on > many lines. Hmm.. but wouldn't reversal detection help those people who are in the UK and want to receive caller ID? I guess you could watch out for the CPE alert tone that's issued after the line reversal, but I have some vague recollection that maybe you need to provide a load so that wetting current can flow or something after the reversal. SIN 242 rings a bell - it's a BT technical document that describes the protocol exactly. All the SINs are available at www.sinet.bt.com Hmm.. reading up, you could probably get away with just monitoring for the alert tone then? I bow to the greater knowledge of the true techies on the list.. I guess European caller ID is one of those big problems, with lots of countries using lots of different protocols - I think some scandinavian countries even use on-hook DTMF.. Easy answer: Get a digital circuit! :-) ___ 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] Why don't I get a ringing sound? - DUH!
I got it! Nothing like posting to the mailing list when you're going to look
stupid to help you find the answer yourself!
The answer is to use waitforring(1)!
Thanks!
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Sunday, May 02, 2004 2:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Why don't I get a ringing sound?
I am using the following macro to dial a ZAP channel. When I dial in, *
answers and I go to voicemail. I never hear any ringing, though. It doesn't
work with the Ringing command before or after the Dial command.
[macro-zapdial]
;
; call a ZAP extension for ${ARG2} seconds, and then voice mail
; ${ARG1} - Extension
; ${ARG2} - Time to ring
exten => s,1,Dial(ZAP/${ARG1},${ARG2})
exten => s,2,ringing
exten => s,3,Voicemail(u$[${ARG1} + 99]) ; match the channel to the mailbox
exten => s,4,Goto(${ARG1},1) ; If they press #, return to start exten =>
s,104,Voicemail(b$[${ARG1} + 99]) exten => s,5,Goto(${ARG1},1) ; If they
press #, return to start
Here is what the log shows:
-- Zap/1-1 is ringing
-- Nobody picked up in 2 ms
-- Hungup 'Zap/1-1'
-- Executing Ringing("Zap/49-1", "") in new stack
-- Executing VoiceMail("Zap/49-1", "u100") in new stack
-- Playing 'vm-theperson' (language 'en') May 2 18:36:45
WARNING[163851]: chan_zap.c:1193 zt_set_hook: zt hook
failed: Device or resource busy
-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (macro-zapdial, s, 3) exited non-zero on 'Zap/49-1' in
macro 'zapdial'
== Spawn extension (main, 100, 1) exited non-zero on 'Zap/49-1'
-- Hungup 'Zap/49-1'
Paul Mahler
[EMAIL PROTECTED]
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Re: [Asterisk-Users] RE: Caller ID
On Monday 03 May 2004 00:02, Paul Crick wrote: > Hmm.. but wouldn't reversal detection help those people who are in the UK > and want to receive caller ID? Yes and no. If we could leave the FSK modem running all the time, then hardware support for the polarity switch wouldn't be as vital. > www.sinet.bt.com Aye, I've poked through that and changed some of the Hz values in callerid.c, but it's getting the code triggered that's the tricky bit! :) > I bow to the greater knowledge of the true techies on the list.. Me too! :) > I guess European caller ID is one of those big problems, with lots > of countries using lots of different protocols - I think some scandinavian > countries even use on-hook DTMF.. I've been looking through the Asterisk source today a lot, and the actual problem with UK CLID seems to be * was designed solely for the USA model of "once we get a ring, despatch a new thread to listen for CLID". So, yes if it were possible to detect the polarity switch, that would be enough to trigger a 'ring' event in *, and then the CLID code could trigger :) > Easy answer: Get a digital circuit! :-) :) You ain't experienced the danger of me with a soldering iron :) gdh ___ 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: Caller ID
On Monday 03 May 2004 00:02, Paul Crick wrote: > Hmm.. reading up, you could probably get away with just monitoring for the > alert tone then? And this time round, I actually read your message. Yes, you're quite right - it might only be a tiny 'bleep' on the line, but if the line is currently not in use, and * is the only thing connected to the line, actually it must be safe to assume that a momentary sound at the alert tone frequency is the announcement for CLID incoming, and hence an incoming call! Cheers, Gavin. ___ 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] Re: Simple SIP X-Lite Configuration Failing
Can anyone help. I've changed the extensions.conf file as follows:
extensions.conf
[sip] ; context for X-Lite Clientsexten =>11,1,Dial(SIP/jay,20,tr)exten =>22,1,Dial(SIP/jtest,20,tr)
I'm still getting the Auto-congesting error (and circuit-busy). Does anyone know what is causing this in such a simple configuration?
localhost*CLI> -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack -- Called 410May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: Auto-congesting SIP/410-a4a1 -- SIP/410-a4a1 is circuit-busy == Everyone is busy at this timeJ Poz <[EMAIL PROTECTED]> wrote:
I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening:
localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtestMay 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congesting SIP/jtest-6a1e -- SIP/jtest-6a1e is circuit-busy == Everyone is busy at this timeMay 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No application 'DialCongestion' for extension (sip, 22, 2) == Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
My setup is very simple and basic:
SIP.conf
[general]port = 5060bindaddr = 0.0.0.0context = sip; Default
[jay]type=friendsecret=jaysipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay <400>"disallow=allallow=gsmcontext=sip
[jtest]type=friendsecret=jaytestsipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay Test <410disallow=allallow=gsmcontext=sip
extensions.conf
[sip] ; context for X-Lite Clientsexten =>11,1,Dial(SIP/jay,20,tr)exten =>11,2,Congestionexten =>22,1,Dial(SIP/jtest,20,tr)exten =>22,2,DialCongestion
Lastly, here's my client setup
Display Name: JayUser Name & Authorization User: jay
Password: jaysip Domain/Realm: 192.168.1.20 SIP Proxy: 192.168.1.20
Display Name: Jay TestUser Name & Authorization User: jtest
Password: jaytestsip Domain/Realm: 192.168.1.20 SIP Proxy: 192.168.1.20
Any help anyone can give me would be appreciated since I've already spent HOURS on this and have made absolutely no progress in debugging this (didn't find anything in any of the archives nor wiki pages).
J...
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Re: [Asterisk-Users] FXO line hum w/ Z-plex 10
You're likely getting what is called a "ground loop". This usually occurs when you have the various pieces of equipment in a system seeing different ground potentials. For example, if your telephone demarc is connected to a short ground rod outside your house, and your electrical power system is grounded to a cold water pipe you'll have different ground references and different ground potentials between the various elements of the system. The first thing to do is make sure all your grounds are all bonded together. That way the same potential exists between all the grounding points. The next thing is to look at the way all the equipment in your system is grounded. Even if all the main ground points are bonded together, you can still get ground loops if one piece of equipment has a different ground connection than the rest. Give this a whirl and let me know if it makes a difference. THX/BDH On Sun, 2004-05-02 at 16:15, Jamin W. Collins wrote: > I've recently begun integrating an Asterisk system into my house. I > purchased the Dev Kit a year or two ago when Digium was selling it as a > Z-plex 10 channel bank with the T100P. > > I've recently found that when I connect the serial monitoring port to my > system it introduces a noticable hum on my incoming FXO lines. Anyone > know (or have suggestions about) how to prevent this? ___ 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: Simple SIP X-Lite Configuration Failing
Sorry for any confusion.But in my latest error, instead of calling my clients "jay" and "jtest", I'm calling them "400" and "410".. Everything else is still the same and it's same problem. My guess is that I've set a parameter incorrectly and therefore Asterisk thinks there's only one client so any calls I try to make between the two fail since it thinks the other client is busy. But I don't understand enough to interpret the error message. I thought the SIP part would be the easy part - I already have the FXO and FXS interfaces working.
Again, thanks for anyone who can help me since I am at a loss!J Poz <[EMAIL PROTECTED]> wrote:
Can anyone help. I've changed the extensions.conf file as follows:
extensions.conf
[sip] ; context for X-Lite Clientsexten =>11,1,Dial(SIP/jay,20,tr)exten =>22,1,Dial(SIP/jtest,20,tr)
I'm still getting the Auto-congesting error (and circuit-busy). Does anyone know what is causing this in such a simple configuration?
localhost*CLI> -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack -- Called 410May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: Auto-congesting SIP/410-a4a1 -- SIP/410-a4a1 is circuit-busy == Everyone is busy at this timeJ Poz <[EMAIL PROTECTED]> wrote:
I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening:
localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtestMay 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congesting SIP/jtest-6a1e -- SIP/jtest-6a1e is circuit-busy == Everyone is busy at this timeMay 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No application 'DialCongestion' for extension (sip, 22, 2) == Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
My setup is very simple and basic:
SIP.conf
[general]port = 5060bindaddr = 0.0.0.0context = sip; Default
[jay]type=friendsecret=jaysipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay <400>"disallow=allallow=gsmcontext=sip
[jtest]type=friendsecret=jaytestsipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay Test <410disallow=allallow=gsmcontext=sip
extensions.conf
[sip] ; context for X-Lite Clientsexten =>11,1,Dial(SIP/jay,20,tr)exten =>11,2,Congestionexten =>22,1,Dial(SIP/jtest,20,tr)exten =>22,2,DialCongestion
Lastly, here's my client setup
Display Name: JayUser Name & Authorization User: jay
Password: jaysip Domain/Realm: 192.168.1.20 SIP Proxy: 192.168.1.20
Display Name: Jay TestUser Name & Authorization User: jtest
Password: jaytestsip Domain/Realm: 192.168.1.20 SIP Proxy: 192.168.1.20
Any help anyone can give me would be appreciated since I've already spent HOURS on this and have made absolutely no progress in debugging this (didn't find anything in any of the archives nor wiki pages).
J...
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[Asterisk-Users] Adit 600 FXO card
I am looking at buying an adit 600 off ebay but the ones listed only have FXS cards in them. Does any one have an FXO card they would like to sell? Thanks -- Jon J. Brandon [EMAIL PROTECTED] http://www.monsoonretail.com ___ 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] Channel Bank - Vina T-1 Integrator
Has anyone tried a Vina T-1 Integrator as a channel bank with Asterisk? They appear to be plentiful, but I want to make sure I'm not buying a brick. THX/BDH ___ 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] module help?
Simple solution on redhat machines In the zaptel source tree (At least the CVS one) there is a file called zaptel.init. This is a script that will allow you to install all needed modules. To use it do this: cd /usr/src/zaptel cp zaptel.init /etc/init.d/zaptel chkconfig --add zaptel chkconfig --level 2345 zaptel on Now every time you reboot all the zaptel modules will be install automatically. PS Why this is not done in the make install script is beyond me. Scott 700-297-0469 - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, May 02, 2004 6:59 PM Subject: Re: [Asterisk-Users] module help? > > > > I've installed the new TDM04B 4-port FXO card and its working. After > > > a reboot, when I do lsmod I see the wcfxo module but not the wcfxs > > > even though both are listed modules.conf. > > > > > > If I "modprobe wcfxs", then lsmod has both modules showing. > > > > why you need wcfxs on a quad-fxo ? > > Because the support folks at digium said on Friday the supporting routines > for the new fxo card are actually in wcfxs. > > > > The wcfxs module is the last one in the modules.conf. Is the order > > > of entries sensitive in modules.conf? > > > > modules.conf != loaded modules. > > as the name suggest, it contains only configuration params > > for modules > > > > > > Do I need to be concerned with wcfxs not showing before starting > > > asterisk? Any suggestions? > > > > sure. > > learn something more about kernel, modules and what > > is modules.conf > > > > bug us with asterisk related questions, not > > with what-are-kernel-modules? questions. > > Okay, then let me reword this just for you. > > Is there a problem with the asterisk "make install" process that > might be considered the root-cause for wcfxs not showing up > in lsmod? > > > > ___ > 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] no dial tone
I just got the X100P and the TDM400P with one module on it, I had installed asterisk and confirgured some file, but I can't get a dial tone on my analog phone. can someone help? Regards Leo ___ 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] module help?
Why copy...use the make command(same with asterisk)... make config Will do all that for you. DP On Sun, 2004-05-02 at 22:32, Scott Weis wrote: > Simple solution on redhat machines > > In the zaptel source tree (At least the CVS one) there is a file called > zaptel.init. This is a script that will allow you to install all needed > modules. To use it do this: > > cd /usr/src/zaptel > cp zaptel.init /etc/init.d/zaptel > chkconfig --add zaptel > chkconfig --level 2345 zaptel on > > Now every time you reboot all the zaptel modules will be install > automatically. > > PS Why this is not done in the make install script is beyond me. > > Scott > 700-297-0469 > - Original Message - > From: "Rich Adamson" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Sunday, May 02, 2004 6:59 PM > Subject: Re: [Asterisk-Users] module help? > > > > > > > > I've installed the new TDM04B 4-port FXO card and its working. After > > > > a reboot, when I do lsmod I see the wcfxo module but not the wcfxs > > > > even though both are listed modules.conf. > > > > > > > > If I "modprobe wcfxs", then lsmod has both modules showing. > > > > > > why you need wcfxs on a quad-fxo ? > > > > Because the support folks at digium said on Friday the supporting routines > > for the new fxo card are actually in wcfxs. > > > > > > The wcfxs module is the last one in the modules.conf. Is the order > > > > of entries sensitive in modules.conf? > > > > > > modules.conf != loaded modules. > > > as the name suggest, it contains only configuration params > > > for modules > > > > > > > > Do I need to be concerned with wcfxs not showing before starting > > > > asterisk? Any suggestions? > > > > > > sure. > > > learn something more about kernel, modules and what > > > is modules.conf > > > > > > bug us with asterisk related questions, not > > > with what-are-kernel-modules? questions. > > > > Okay, then let me reword this just for you. > > > > Is there a problem with the asterisk "make install" process that > > might be considered the root-cause for wcfxs not showing up > > in lsmod? > > > > > > > > ___ > > 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] X101P problems
All, I'm having some issues with an X101 clone. The machine is not plugged into the network right now, but I'll pull the configs and send them on shortly, but they are similar to the sample configs. Problem 1: "ZAP cannot create channel" For some reason, the cards hang after a call. It's pretty annoying since the only thing that will fix it is a restart of asterisk. There are two cards in the machine, both with their own, non-overlapping IRQ's. ztcfg and zap show channels show both cards properly configured and working. Inbound calls work properly, but outbound PSTN calls hang the interface. Here is the strange part, and the two problems may be related. The way outbound calls are originated is through a Sipura SIP adapter and an analog phone. During the call, as monitored at the command line, * shows the SIP call originating, the dial command for the ZAP interface being started and _the ZAP interface answering_, which seems to result in the originating station not getting a ring tone... After the call, it shows the SIP call terminated, but the ZAP channel remains unavailable until a * restart. Reloading the configs does nothing. Problem 2: "No ring tone on SIP to PSTN calls" When calling a PSTN phone from a SIP phone, there is no ring tone. Putting an 'r' at the end of the dial line results in a single ring only, under every circumstance. The * montering shows that the ZAP interface answers the call, which is wierd. Perhaps this has something to do with context? Should outbound calls be in a different context than incoming calls? Is it possible that * is answering itself? This would seem to be wrong as the remote number (my cell in this case) actually rings normally, there's just silence on the * side... Perhaps someone can shed light on my extensions.conf when I manage to get to it Problem 3: "Random ringbacks" I'm getting random ringbacks and just generally random rigs on the Sipura SIP adaptor. These seem to occur when the connection between * and the Sipura is disrupted, as well as at random times. Is there a way to fix or disable this behavior? It's very annoying. If anyone can shed any light on these problems, I'd be gratefull. I almost threw the box out the window today I was so frustrated... Thx. Chris. -- chris maresca senior partner - www.olliancegroup.com linux, up 27 days ___ 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] Re: Adit 600 FXO card (Jon Brandon)
Whenever we have needed any kind of Carrier Access (Adit) equipment, we have used Suntel Data. We found them on eBay and have been buying from them ever since. They have always been good about fast shipments and provide good used pricing on equipment. http://www.sunteldata.com/ Brian ___ 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] Adit 600 FXO card
$400 new in stock - Original Message - From: "Jon Brandon" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, May 02, 2004 8:41 PM Subject: [Asterisk-Users] Adit 600 FXO card > I am looking at buying an adit 600 off ebay but the ones listed only have > FXS cards in them. Does any one have an FXO card they would like to sell? > > Thanks > > > -- > Jon J. Brandon [EMAIL PROTECTED] http://www.monsoonretail.com > > ___ > 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 goes international :-)
Hi, I'm from Spain and I have developed in Perl 2 scripts to say_number and say_digits in 3 diferents language (spanish, german and english). The problem is that I don't know how to adapt it to C in order to complement say.c If somebody can help me, I will be very pleasured. Thanks a lot. - Mensaje Original - De: "Olle E. Johansson" <[EMAIL PROTECTED]> Fecha: Jueves, Abril 29, 2004 8:49 am Asunto: Re: [Asterisk-Users] Asterisk goes international :-) > Altus Snyman wrote: > > > So what do I have to do to add South-Africa to this list? > > > If you are saying numbers in a different way than english or if > you are thinking > about another language, check the latest version of say.c in CVS > head and > see if you can construct the syntax needed for your language. > > When you have a patch, open a bug report under > "Internationalization" in > http://bugs.digium.com and add the patch there. > > I don't know how Asian languages work, but it would certainly be > interestingto see patches for those. > > Also, we need a larger group that works with the next generation > architecture. > /O > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinf > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinf > ___ 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] * Newbie installation advice
Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the wildcard?) Chanel Bank: * Adit 600 3FXS, 1FXO * We will start with 6 PSTN lines Phones: * Aastra PowerTouch 480 (Management, Customer Service etc) * Aastra Meridian 8004 (break room, warehouse floor etc) * Reception? Any comments or suggestions would be appreciated as I have no idea what type of phone to give reception. Reception typically has a multi-line phone to answer incoming calls. * Polycom SoundStation 200 EX (Conference room speaker phone) Am I missing anything? I see from the archives that a lot of people have used the PowerTouch phones. What do people think of them? Are there other ADSI phones that are better or just as good for less money? Thanks -Jon -- Jon J. Brandon [EMAIL PROTECTED] http://www.monsoonretail.com ___ 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 echo-free and trouble-free system?
On Fri, Apr 30, 2004 at 10:28:02AM -0500, Barton Hodges wrote: > [EMAIL PROTECTED] wrote: > > The real problem arises when : > > - you have some echo induced somewhere (your call goes through a 2 > > wire line) > > - you have some delay induced somewhere (you use VoIP for instance) > > Following the "2-wire to 4-wire causes echo" thought, the following > should not result in noticable echo, true? > > - Analog phone <-> TDM10B-FXS <-> Asterisk <-> TDM01B-FXO <-> PSTN Yes, because although echo will exist, delay should be short enough so that you don't notice. Never tried such a setup myself, though. Please furthermore note that Asterisk uses "pseudo" TDM. In real telco world, PCM highways that interconnect trunks and devices switch one byte every 8000th/sec. OTOH, Zaptel devices switch eight bytes every 1000th/sec. This is due the to PC bus architecture (it would cause way too much overhead otherwise). So the delay is actually 8 times longuer (at least) than in the PSTN. > - VOIP Phone <-> Asterisk <-> VOIP Phone > - VOIP Phone <-> Asterisk <-> T100P <-> PRI > Nobody's supposed to generate echo on VoIP phones. However, the "PRI" side will probably connect to a 2-wire PSTN set at the remote end, so you will get echo from there. > However, the following could result in noticable echo (as I am > experiencing): > > - Analog phone <-> ATA <-> Asterisk <-> TDM01B-FXO <-> PSTN > - VOIP Phone <-> Asterisk <-> PSTN > Definetly. Although Asterisk (zaptel, actually) make a fairly good job at cancelling it. > What about the following as described in Raymond McKay's setup (Thank > you Raymond) > Does the channel bank provide the needed, and adequate echo > cancellation? > I don't have any experience with channels banks. Not very common stuff in Europe. > > Since you state that echo cancellation needs to be performed closest > to the source, could the Grandstream HandyTone-286 be doing an > inadequate job of echo cancellation? If this is the case, does anyone > have experience with another ATA (Sipura SPA-1000, Cisco ATA-186, > etc.) that does such a great job of echo cancellation, that the > "2-wire to 4-wire" situation is not an issue? Does Grandstream have > improved echo cancellation scheduled for a future firmware upgrade? > The Sipuras are definetly better than the HandyTones. I've heard that the forthcoming GS firmwares will enhance echo cancellation performance, though. -- Nicolas Bougues Axialys Interactive ___ 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] GrandStream 1.0.4.55 Firmware
On Fri, Apr 30, 2004 at 06:13:49PM +0100, Senad Jordanovic wrote: > [EMAIL PROTECTED] wrote: > > Go to 1.04.54. This is pretty stable. Find it at > > www.telappliant.com/grandstream > > > Does this version supports TFTP auto configuration? If it does, please > contact me off the list for volume purchase discussion! > Err, all (1.0.4.x at least) GS firmwares support TFTP autoconfiguration ! -- Nicolas Bougues Axialys Interactive ___ 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] * Newbie installation advice
On Mon, 2004-05-03 at 01:06, Jon Brandon wrote: > Hello, > > I'm about to install asterisk as the PBX at a location that my company has > just moved into and I would like to get some comments and advice on the > installation. I am new to * and don't want to make any big mistakes so I > would love to hear whatever anyone has to say. Your first mistake _may_ be the rush to learn. Good experiences normally require you to have a time to get used to the application and PSTN problems before you attempt to go through a roll out. > Here is what I have so far > Server: > * 2.8Ghz P4 - 1G ram > * T400P Tormenta II (is this as good as the wildcard?) > Chanel Bank: > * Adit 600 3FXS, 1FXO > * We will start with 6 PSTN lines If you are going to start with 6 lines, you should decide how soon you might upgrade. You then should look into the cost difference to get either channelized T1 or PRI. You will be much happier with a T1 than analog lines. Specifically look at how many people here fight with echo, a T1 makes the risks of echo lower. It also becomes cheaper as the number of lines go up than analog lines. At some point in your growth, if you continue with analog lines, the telco will drop a similar Adit right next to yours to break the T1 they bring in out to the analog lines you order. > Phones: > * Aastra PowerTouch 480 (Management, Customer Service etc) > * Aastra Meridian 8004 (break room, warehouse floor etc) > * Reception? Any comments or suggestions would be appreciated as I > have no idea what type of phone to give reception. Reception typically has > a multi-line phone to answer incoming calls. Multiline isn't necessary. For that matter, a receptionist isn't overly necessary. You can help direct callers to a extension pretty easy with a menu system you script. Then you just need to designate a couple of people/phones that are used in the case the caller refuses to follow the menu or doesn't find the person they are looking for. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] South-Africa
My advice is just sell them. no-one I know is bothered with Icasa approval, as long as it works, its fine. That card has FCC approval, as far as I know. ALles van die beste! Regards Clive On Fri, 30 Apr 2004 15:17:13 +0100 WipeOut <[EMAIL PROTECTED]> wrote: > Altus Snyman wrote: > > >Good day all > >I'm in South-Africa,currently we are using openline4 > cards for our pbx > >systems.Now we first need approval on the cards form > icasa(a government > >standards) before we can use the card.The market here is > very big for a > >system like asterisk.The only problem is to get a card > approved(for a > >small company like us) its just about impossible. > >Now what I'm looking for is a company that will import > an approve a card > >or if someone out of South-Africa now of such a card? > >The market is very big here > >Let me Know > >Thanks > >Altus > > > > > Just don't tell anyone.. ;) > > We tried getting Modems approved in SA about 8 years ago > and in the end it just wasn't worth it.. The regulators > were a joke and their costs were rediculous.. It may have > improved now.. > > Good luck.. > ___ > 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 _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ 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
