RE: [Asterisk-Users] Softfax/spandsp Makefile.patch rxfax/txfax
Search the mailing lists, this has been answered a million times. Edit the Makefile and remove the entries for both app_rxfax.o and app_txfax.o and it will compile fine. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Saturday, June 19, 2004 10:14 PM To: Asterisk Mailling List Subject: [Asterisk-Users] Softfax/spandsp Makefile.patch rxfax/txfax I followed the instructions at http://www.opencall.org/instructions.html and http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html I was able to compile spandsp (./configure ; make ; make install), manually patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the supplied patch does not fit the actual CVS apps/Makefile After make clean ; make install, I received this error: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init':../include/asterisk/lock.h:214: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:214: error: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:214: error: for each function it appears in. Any help? (yes, I copied app_rxfax.c app_txfax.c app_dtmftotext.c to apps dir also) ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] asterisk -rx not working well
It's exiting before the output finishes printing, it's a known bug and a timing issue There's a patch I put that's a hack to put in a timeout for exit, you should be able to find it on bugs.digium.com Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephan Wik Sent: Saturday, June 19, 2004 1:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk -rx not working well We're trying to use asterisk -r -x sip show peers to monitor sip phone availability. Sometimes the command shows the correct output but 9 times out of 10 all that is returned is: Name/usernameHost Mask Port Status with no listing. Anyone else see this behaviour? Stephan ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] asterisk -rx not working well
The problem is that it uses a socket, and doesn't KNOW when the end of the command is. Somebody else IS working on fixing this so there is a control connection and it will know. It does finish writing everything that the remote console got from the * server before it exits... It just hasn't got everything yet. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Saturday, June 19, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk -rx not working well On Sat, 2004-06-19 at 07:05, Sam Bingner wrote: It's exiting before the output finishes printing, it's a known bug and a timing issue There's a patch I put that's a hack to put in a timeout for exit, you should be able to find it on bugs.digium.com Instead of timing the exit, why don't you just flush the STDOUT and STDERR file descriptors? Seems this could be done before exit no matter what the reason was. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephan Wik Sent: Saturday, June 19, 2004 1:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk -rx not working well We're trying to use asterisk -r -x sip show peers to monitor sip phone availability. Sometimes the command shows the correct output but 9 times out of 10 all that is returned is: Name/usernameHost Mask Port Status with no listing. Anyone else see this behaviour? Stephan ___ 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 -- 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] Re: Fedora2 and Kernel 2.6 again!
That would be correct, I posted it to list a few times, but my wiki account is broken (bleh) so I never posted it there Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mario Velasco Sent: Wednesday, June 16, 2004 9:19 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Fedora2 and Kernel 2.6 again! I'm a newbie, but I found this information that works for Fedora2: You need to make a symlink /usr/src/linux-2.6 ... you can reference it to /lib/modules/2.6.5-1.358/build/ (I think you don't need to compile your kernel for this)... or you can reference it to /usr/src/linux-2.6.5-1.358/ (but you need to compile your kernel) There is additional information at README.linux26 Mario Velasco _ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.yahoo.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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] spandsp w/libtiff-3.6.1?
Did you actually look at that patch? --- It fixes some bug in 3.6.1 related to faxing... If so, sorry for wasting all your bandwidth :b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Monday, May 31, 2004 1:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp w/libtiff-3.6.1? Aaron J. Angel wrote: Has anyone used spandsp with a patched libtiff 3.6.1 successfully? http://bugs.hylafax.org/bugzilla/show_bug.cgi?id=500 Of course not. That is why I keep telling people not to use it. :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] spandsp wont compile.
You shouldn't put /usr/include in ld.so.conf, needing to do so means you have something installed wrong... And I've never heard of anything getting installed that wrong ;) Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Sunday, May 30, 2004 1:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] spandsp wont compile. Yes, success! I deleted the tiff libs I had and installed ver 3.6.0 and was able to compile and load the application modules. Now I just have to do some tweaking and t-shootin' in ext.conf. Thanks and a Shout Out to all for their advice and help. Couldn't have done it w/out you. I also had to put /usr/include in ld.so.conf. Hope this helps others. On Sat, 2004-05-29 at 18:09, Mark Musone wrote: Your most likely compiling against one tiff library version, but loading up another... Do a: ldd app_rxfax.so to see what tiff library it's compiled against, and then also try to find all the places where libtiff is on your machine and remove the incorrect one.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Saturday, May 29, 2004 6:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] spandsp wont compile. /etc/ld.so.conf /usr/X11R6/lib /usr/lib/qt3/lib /usr/local/libUnable to load module app_rxfax.so May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize /usr/local/lib/libtiff /usr/lib/asterisk/modules the mods compiled BUT now won't load. On Fri, 2004-05-28 at 23:25, Todd Lieberman wrote: add /usr/local/lib to your /etc/ld.so.conf Then run ldconfig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ 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 ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] spandsp wont compile.
Add the path to it to /etc/ld.so.conf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Friday, May 28, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] RxFAX generates no tiff file
You should Answer() your calls... In the 5000 exten, you could move your Answer to after the dial if you like... And the h exten hangs up if it doesn't exist so that's redundant, but not bad Sam [internalexten] exten = 5000,1,Answer() exten = 5000,2,Dial(SIP/mike,60,tr) exten = 5000,3,SetLanguage(de) exten = 5000,4,Playback(vm-nobodyavail) exten = 6000,1,Answer() exten = 6000,2,WaitMusicOnHold(30) exten = 7000,1,Answer() exten = 7000,2,rxfax(/tmp/testfax.tif) [default] include = internalexten exten = h,1,Hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Heininger Sent: Sunday, May 23, 2004 1:41 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Am 23.05.2004 um 04:33 schrieb Steve Underwood: How do you run rxfax? You problem is probably something to do with that. Your's is the first report I have had of no TIFF file whatsoever. [internalexten] exten = 5000,1,Dial(SIP/mike,60,tr) exten = 5000,2,SetLanguage(de) exten = 5000,3,Playback(vm-nobodyavail) exten = 6000,1,WaitMusicOnHold(30) exten = 7000,1,rxfax(/tmp/testfax.tif) [default] include = internalexten exten = h,1,Hangup The context for the inbound call is [default] and goes to extension 7000. TIA, Mike ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] extension pattern matching
I think may be able to do that with _[a-z][a-z]. But I haven't tried it, you need to use 2 to make sure you don't overwrite the system extensions. As I understand the * regex implimentation, you can't do _.[a-z]. to match any letters in dialplan anywhere, but that is what you really wanted I think. You could always code it in ;) Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Graham Turner Sent: Sunday, May 23, 2004 9:09 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] extension pattern matching dear all, was hoping someone could give me instruction on the syntax of extension pattern matching for letters the proposed 'dial plan' is one where any letter in the dialled digits causes the pbx to assume we are dilaling a sip url and as such forward to the appropraite sip service provider was hoping to avoid the plan in john todd's example that assumes anything prefixed with 3 is a sip address gt ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] ZAPTEL not loading on FC2
Change your symlink to not point to the linux source tree, but rather point at /lib/modules/2.6.5-358/build, and just do a make linux26 Or apply this patch to your makefile... Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Taz Man Sent: Sunday, May 23, 2004 4:57 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZAPTEL not loading on FC2 Hello all, I've just installed the Fedora core 2 and tried to compile the asterisk and the zaptel drivers Asterisk went smooth but I had troubles with the zaptel. I did copy the .config file under the kernel source and make oldconfig and make include/asm ; make include/version.h ; make SUBDIRS=scripts I was able to compile the zaptel source, using the make linux26 and then ran make install. I see that the .ko files are in place (under /lib/modules/2.6.5-1.358/misc/...) but when trying to load the zaptel I get: [EMAIL PROTECTED] misc]# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko): Invalid module format [EMAIL PROTECTED] misc]# modprobe wcfxo WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko): Invalid module format FATAL: Error inserting wcfxo (/lib/modules/2.6.5-1.358/misc/wcfxo.ko): Invalid module format FATAL: Error running install command for wcfxo uname -a gives Linux server 2.6.5-1.358 #1 Sat May 8 09:04:50 EDT 2004 i686 i686 i386 GNU/Linux any help? any Ideas? 10x, Ronen ___ 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 makefile-kernel26.diff Description: Binary data smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
Really you should link /usr/src/linux-2.6 to /lib/modules/`uname -r`/build then you don't have to do anything special and it'll build... That directory and all the files in it are installed by the kernel rpm, you don't even need kernel-source for it... Although I haven't tried compiling without it installed I patched my zaptel Makefile to just reference that directory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua M. Thompson Sent: Thursday, May 20, 2004 5:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6 On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. -- Joshua M. Thompson [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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] x100p / Answer- Flash - Dial
Title: Message Even if you could get that to work properly, which I dont know... the callprogress detection is horrible; if you want to do that reliably you need a T1,ISDN or IPinterface to the switch (something that actually provides proper call progress) Sam -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan FernandezSent: Saturday, May 08, 2004 11:44 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] x100p / Answer- Flash - Dial I have an X100P connected to an extension of aPanasonic PBX.When a call from the PSTN comes in,it is routed directly to theextension where the x100p is .I want* to answer the call, play amessage and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then todialan IAX channel. The problem is that when I do a Flash on thezap channel, and then try to dial a new extensionvia that zap channel I get the following error "can't createzap channel". If I do a SendDTMF()thecalldoes get transfer to the new extension but then * gets out of the callloop and don't know it is answered or not by the new extension. AmI missing something? Why am I getting the "can't creatza channel" Thanksin advance. Dan smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] Fax problem
Use ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela Sent: Friday, April 23, 2004 7:52 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fax problem Hi, We have a machine with an *'s with Digium TDM400P and connected wit other machine with *'s an TDM400P too. Well, I have a fax connected to each machine, and the protocol in the middle is IAX2 alaw. The fax between two fax, on in each machine, not work. The fax answer, but error in comm. Which can be the problem ?. What can I do to find the problem ? Thanks, in advance, Pedro ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] libspandsp.so.0
It's worked good for me... Only had a garbled page once when it was a 15 page fax, and that was a few versions ago so I'm not sure if it would do the same now or not Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Enger Sent: Sunday, April 18, 2004 5:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] libspandsp.so.0 Put it in /etc/ld.so.conf (the path to the lib dir the file is in) then run ldconfig. Has anyone had any success with rxfax? Every time I have used it the tiff file has a garbled page. On Mon, 2004-04-19 at 11:02, Karl Brose wrote: ldconfig - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, April 18, 2004 19:24 Subject: [Asterisk-Users] libspandsp.so.0 I successfully compiled installed the spandsp-0.0.1k.tar.gz modules for faxing and patched the asterisk according to the readme and rebuilt and installed * but I am getting this error when attempting to start *. The libspandsp.so.0 file exists and I have coppied it to several directories recompiled and have the same results. What am I doing wrong? help please [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: loader.c:239 ast_load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory Apr 18 18:57:20 WARNING[1024]: loader.c:407 load_modules: Loading module app_rxfax.so failed! ___ 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 -- Matthew Enger [EMAIL PROTECTED] Mob: 0412 463 080 Direct: (03) 9747 4001 X Integration A Netcruiser Pty Ltd business Ph: 1300 730 997 Fax: 1300 136 720 -- Matthew Enger [EMAIL PROTECTED] Xintegration ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] Upgrade firmware on iaxy?
If you have a new enough version of the IAXy firmware on the IAXy, then it will automagically be upgraded as soon as * sees it has an old firmware (via the IAX protocol) --- if you don't have a new enough version, digium has to do it by what I've heard Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Wednesday, April 14, 2004 10:10 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Upgrade firmware on iaxy? I've googled and grepped myself silly. I see the iaxy.bin file there in the contrib tree of the asterisk source, but nowhere have I been able to find out how to get it sent to the device. . . Anyone know? 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
* listens for fax tones as soon as you Answer() the line. If you Answer
the line before ringing the local lines, it will actually detect fax tones
while in the Dial statement.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird
Sent: Sunday, March 28, 2004 5:52 PM
To: Martin List-Petersen
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
On Mar 28, 2004, at 7:40 PM, Martin List-Petersen wrote:
; I'm using a shared analog line for testing this, so I'm using the
fax
; autodetection code to yank faxes out of my IVR and into the 'fax'
; pseudo-extension
[outside]
...
exten = fax,1,Goto(fax,2201,1)
I would be interested in how you do fax autodetection.
I don't do anything particularly special, Asterisk just makes it work.
This is using a bog-standard POTS line at home. Here's the relevant
part of my config:
[macro-outsideline]
exten = s,1,LookupCIDName
exten = s,2,SetMusicOnHold(random)
exten = s,3,Dial(${PHONES},13,Ttm)
exten = s,4,Answer
exten = s,5,Goto(outside-ivr,s,1)
[outside-ivr]
; This is the outside IVR
; Playback a We're not home message
; To leave a message for Scott, press 1
; To leave a message for C, press 2
; Otherwise stay on the line.
;
; Also, 3 = main voicemail
; 4 = check voicemail (main)
; 5 = check voicemail
; 6 = DISA (with password)
;
; Check for fax, too
exten = s,1,NoOp
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(2)
exten = s,4,Wait(1)
exten = s,5,Background(laird/ivr-greeting)
exten = t,1,VoiceMail(s2201)
exten = t,2,Hangup
; other stuff goes here, but it's not really important
exten = fax,1,Answer
exten = fax,2,Goto(fax,2201,1)
[outside]
exten = s,1,Macro(outsideline)
exten = fax,1,Goto(fax,2201,1)
95% of this isn't important for faxing, but I included it for context.
The big issue is the IVR stuff and the 'fax' extension. Once we get to
the IVR, asterisk is listening for DTMF tones and apparently also fax
tones. If it hears a fax, then it goes to the 'fax' extension. That's
it.
Scott
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RE: [Asterisk-Users] Error installing/compiling cdr_mysql addon
You need to install the mysql-devel rpm if you use redhat Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Sunday, March 28, 2004 2:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error installing/compiling cdr_mysql addon When I try to compile the cdr_mysql addon, I get the following error: [EMAIL PROTECTED] asterisk-addons]# make cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 I have MySQL installed and have tested it - it is working, I can create databases etc. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ 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 smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] G.729 quiestion
If you buy the codec, it will do conversion... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of NetOne Administrator Sent: Friday, January 16, 2004 3:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 quiestion Hi all! If i purchase the G.729 codec for *, can Asterisk use it for convertion, or just pass-through only? I need to be able to convert from G.729 to iLBC (or GSM maybe) and vice versa. Is it possible with *? Greetings, Doichin Dokov NetOne - Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] Re: 911 and lawsuits and redundancy
Also, if you ONLY run * on the system, you can lock it down so that the security bugs are pretty much non-exploitable... Ipchains/etc. You don't even HAVE to run ssh or any remote management if you want to to be just like a regular PBX system Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, January 07, 2004 5:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy I need to update my Asterisk server that runs all my phones inorder to install a kernel update that fixes a security bug. This is something I would consider happening on a regular basis with a voip enable system, whereas the traditional system might sit in a closet for 10 years never being touched. Let's say I don't want to stay at work until 2 am to reload the system when noone is there. How would you configure and * system(s so that you could take a system offline during working hours without taking out all or parts of the system? I don't use SIP for one, and I forward udp/4562 to my * box from the firewall... of course now what happens if you have to upgrade the firewall? :-) I think the basic solution is a cluster. That way you can upgrade in pieces without losing functionality. There are T1 monitoring/switching devices which will let two boxes share a single T1 and fail over immediately, although you'll lose the calls in progress. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] Port density: DS3 cards?
Well, we know that we would be able to handle a partial DS3... assuming such a thing is possible. Wouldn't people prefer a partial DS3 for say... 12T1's to no way to do that many? Why not just try to get the card working, then testing would show exactly how much data could be handled... Actually, we should be able to get a pretty good idea on that by using two gigabit interfaces and VoIP? Sam Quoting Andy Hester [EMAIL PROTECTED]: I talked to Imagestream this morning about the possibilites. Their lead engineer said that there would be no way to do voice over their DS-3 cards using software processing because it would take too much processing power. It would be possible to do some custom design for their boards that incorpotates hardware processing, but he doesn't know of anything currently available. So unless there's something I/he missed, I guess the answer is no on the DS-3. Andy I have no reason to disbelieve this report, but I will offer some minor scepticism at this reply. A well-equipped PC can currently handle 8 T1 channels, and it seems that only the IRQ issue is causing more channels to not be viable in the current TE410P environment. It would seem reasonable to think that a very well equipped PC (4-way, 8-way?) would be able to handle the processing power requirements of a DS3, whatever was meant by that statement. Of course, there may be other underlying issues specific to ImageStream that make this impossible; I don't know. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users The guy did leave open the possibility that he could be wrong, and said that he'd be glad to answer any further questions or if we had some other way of doing it. If you or some of the others think that this should be possible then perhaps we could get together a list of more specific questions to ask. Thoughts? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX connected to a TDM400 card port
I have exactly that configuration, and it's working fine for me. I have the following config which may or may not be relevant to fax working... echocancelwhenbridged = no callprogress = no I've heard that there have been echo problems when ring and tip are reversed... but mine havn't ever been reversed so I can neither confirm nor deny this. Sam Quoting Dan [EMAIL PROTECTED]: Hi, I have a FAX machine connected to a TDM400 card FXS port. When I receive a fax call through X100 and transfer it to that extension, the FAX machine display REC, but nothing happen (no fax received). There is something special to be done for this configuration? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3 streams for MOH: idea
I have a working MP3 decoder in a thread, using libresample and libmp3lame, but I'm not really happy with it yet Not sure about the legalities but if anybody wants to try this work in progress just let me know Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sharp Sent: Saturday, September 06, 2003 4:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MP3 streams for MOH: idea [thread change, different topic] is How about a little tiny program that connects to a remote host, grabs the contents of an MP3 stream, and pushes it into a FIFO locally? It would be a raw TCP-to-FIFO stream, so mpg123 would be able to digest it as if it was a local file. The program would take two arguments: remote hostname/IP and port, and then the file to which the output would be sent. I don't know how mpg123 handles blocking... Is there any particular reason (rather than not having time to code one and embed it into *) why we can't have our own in-thread connection to an MP3 stream or file, rather than spawning off a process (fork() is expensive as compared to pthread_create()) of mpg123 to play the stream/file? It seems that this spawning/hoping the process dies cleanly is a thorn in a few people's side. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] Music on hold - multiple formats
I have made a patch that uses sox instead of mpg123 to playback music on hold. Sox, when compiled correctly will support mpg, ogg, wav, gsm and numerous other formats. Attached is a diff file that will make change asterisk's behavior to use sox via the perl wrapper I made. To use this patch, you require sox. If you want to be able to play mp3's you will have to use a newer version than that distributed with redhat (12.17.4+), as well as -- you can download the versions I compiled at: http://www.bingner.com/asterisk/sox-12.17.4-1.i686.rpm http://www.bingner.com/asterisk/lame-3.93.1-1.i686.rpm And the following I installed, but didn't need to compile myself: http://dag.wieers.com/packages/libmad/libmad-0.14.2b-2.dag.rh90.i386.rpm If you install all 3 of those I know you can play MP3 files. Additionally, the wrapper watches for asterisk, and if it no longer sees asterisk running it will die. This simplifies life for asterisk. Mark, if you think this patch is stable feel free to apply it... You have my waiver already. Sam moh_sox.diff Description: Binary data smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] VoiceMail recording dialtone
I don't understand how that would affect the voicemail recording dialtone when the phone never rang? 1, User calls 2, Nobody answers in 20 seconds 3, greeting is played (user hangs up somewhere in here, close to end) 4, voicemail is called, and records a dialtone -- phone never rings again here -- Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Thursday, June 19, 2003 11:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMail recording dialtone Well experiment yourself with the code. in wcfxo.c /* Don't accept a ring for another 1000 ms */ wc-ringdebounce = 1000; Try a diffrent value (e.g. 3000 for 3 sec) and in zaptel.h #defineRING_DEBOUNCE_TIME 500 /* 500 ms ring debounce time */ try the same value as in wcfxo.c recompile/reload and test regards Martin On Thu, 19 Jun 2003, Sam Bingner wrote: Zaptel was the version from about 4 days ago when I sent this message, I updated again yesterday night Sam Quoting Martin Pycko [EMAIL PROTECTED]: How old is your zaptel code ? Mark recently increased some timer for that. Martin On Wed, 18 Jun 2003, Sam Bingner wrote: I have an extension setup with voicemail, for incoming calls on an X100P card. It quite often will record about 15 seconds of dialtone... I'm guessing that it picks up the line after the outgoing line has been disconnected. Has anybody else run into this problem? Shouldn't chan_zap be detecting the hangup and ending the connection? Sam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] VoiceMail recording dialtone
OK, I tried upping it to 2000.. See if that changes anything I still don't understand why it would end up directly in voicemail if it picked the line back up instead of calling extension s again if the telco's hangup signal was interpreted as a ring? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Thursday, June 26, 2003 5:24 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoiceMail recording dialtone Unless your telco signals hangup with a dialtone . it should help. The thing is that most propably your X100P hangs up and then picks up the line due to something ... that was my original idea. Martin On Wed, 25 Jun 2003, Sam Bingner wrote: I don't understand how that would affect the voicemail recording dialtone when the phone never rang? 1, User calls 2, Nobody answers in 20 seconds 3, greeting is played (user hangs up somewhere in here, close to end) 4, voicemail is called, and records a dialtone -- phone never rings again here -- Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Thursday, June 19, 2003 11:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMail recording dialtone Well experiment yourself with the code. in wcfxo.c /* Don't accept a ring for another 1000 ms */ wc-ringdebounce = 1000; Try a diffrent value (e.g. 3000 for 3 sec) and in zaptel.h #defineRING_DEBOUNCE_TIME 500 /* 500 ms ring debounce time */ try the same value as in wcfxo.c recompile/reload and test regards Martin On Thu, 19 Jun 2003, Sam Bingner wrote: Zaptel was the version from about 4 days ago when I sent this message, I updated again yesterday night Sam Quoting Martin Pycko [EMAIL PROTECTED]: How old is your zaptel code ? Mark recently increased some timer for that. Martin On Wed, 18 Jun 2003, Sam Bingner wrote: I have an extension setup with voicemail, for incoming calls on an X100P card. It quite often will record about 15 seconds of dialtone... I'm guessing that it picks up the line after the outgoing line has been disconnected. Has anybody else run into this problem? Shouldn't chan_zap be detecting the hangup and ending the connection? Sam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
Re: [Asterisk-Users] VoiceMail recording dialtone
Zaptel was the version from about 4 days ago when I sent this message, I updated again yesterday night Sam Quoting Martin Pycko [EMAIL PROTECTED]: How old is your zaptel code ? Mark recently increased some timer for that. Martin On Wed, 18 Jun 2003, Sam Bingner wrote: I have an extension setup with voicemail, for incoming calls on an X100P card. It quite often will record about 15 seconds of dialtone... I'm guessing that it picks up the line after the outgoing line has been disconnected. Has anybody else run into this problem? Shouldn't chan_zap be detecting the hangup and ending the connection? Sam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMail recording dialtone
I have an extension setup with voicemail, for incoming calls on an X100P card. It quite often will record about 15 seconds of dialtone... I'm guessing that it picks up the line after the outgoing line has been disconnected. Has anybody else run into this problem? Shouldn't chan_zap be detecting the hangup and ending the connection? Sam smime.p7s Description: S/MIME cryptographic signature
