[asterisk-users] linksys spa3102 for faxing
Hi, I have been considering a purchase of the linksys spa3102 for a couple hours but I would like to know from someone here, wether this device will support faxing on my local asterisk server, I have had success sending and recieving faces with an x100p, and recall that in the old documentation, they mention that if I send/recieve faxes, that it all should be done on the local server for best performanc, so Im asuming tha this device may apply because there will be an ethernet cable between the FXO and the asterisk server? thanks! Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! http://tv.yahoo.com/collections/3658 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] basic 3+ way conference call on plain old phones
hi guys, is it possible to do a basic 3-or-more-way conference call when the phones dont support it? I am fully aware of this concept on expensive phones like this one: Grandstream GXP 2000 -Conference call 3-way http://www.youtube.com/watch?v=hlZ6JqE1MT4 The problem is that the basic plain old commercial PBX supports 3-way calling in ugly old phones like this one: http://www.neo-shop.com/tiendas/0009/varios/telefono%20TEIDE-1.jpg connected to an ata like this one: http://www.egk.com.ar/imagenes/hardware/sipura2.jpg The idea is to be caller (A): dial calle (B), once (B) answers press on HOOK or something else to send them to MOH, then dial callee (C), talk to him a little too, then press the same HOOK or something else and the 3, (A)(B) and (C) in a conference call. Unlike the grandstream, this would definitelly have to be done by *, isnt this part of the basic functionality like voicemail that is already done and a couple lines in the config files it will work on all phones done by *? if not, then, how do you recommend me to it? the closest I have seen to shat I am looking for is http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro is there a better alternative? any thoughts? thanks a lot! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] xten will not send tones to * and i from sip phone
hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some sort of configuration out there to tell the xten softphone to work as expected? thanks! Then another problem! I used the i extension, plus _X and _X. to make sure I catch everything that is not propperly dialed. If I take the regular phones that are connected through the sipura ata, then dial 'exten = 700,1,Goto(default,s,1)' so that I get the asking for an extension to reach, I dial a wrong number and walla, its caight by one of my magic numbers! BUT, if I pickup the same phone, and just dial the same wrong number? I just get a busy signal! and there is nothing registered at the CLI even though I added DEBIG to the configuration! :s What can I do to make sure I always send an error sound and never again a busy signal? thanks! Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] muscionhold error message
hi there guys! how can I eliminate this message? [May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player [May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' This is on debian etch 4.0 asterisk 1.4, it happens quite often everyday and I have to scroll a lot to try to find other error messages. btw can I just put some musica wav files in /var/lib/asterisk/mohmp3 ? that would be great to leave asterisk's processor alone thanks! Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Catch all undefined numbers to play a nice message and restart
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like
You can search for the word irc to see my comments,
the line above is my latest unsuccessful test, thanks!
;
;
;
;
;
; begin extensions
;
;
;
;
;
;
[general] ;
language=es
; autofallthrough=yes
clearglobalvars=no
[globals]
; Definiendo variables para usarlas a traves de todo
el
; MINOMBRE=mailinator.net
; MITELEFONOFXO=
; OPERADORA=
;
; Si static esta en no, u omitido, entonces pbx_config
va a sobreescribir
; a este archivo cuando se cambien las extensiones.
Recuerda que todos los
; comentarios de este archivo desapareceran si pasa
eso.
;
; XXX Todavia no ha sido implementado XXX
;
static=yes
;
;
; si stati=yes y writeprotect=no, tambien puedes
guardar al dialplan con
; linea de comandos ejecutando 'save dialplan' y
borrando estos comentarios
;
writeprotect=yes
CONSOLE=Zap/1 ; pendiente entender *
TRUNK=Zap/1 ; Trunk interface *
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0) *
;
; Trunks
;[context] ;exten =
someexten,priority[+offset][(alias)],application(arg1,arg2,...)
[trunkint] ; International long distance
through trunk
exten = _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld] ; Long distance context
accessed through trunk
exten =
_901ZX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]; Local eight-digit dialing
accessed through trunk interface
exten =
_9ZXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
llamada local comun y corriente
exten = _90ZXS0,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
; 020, etc
exten = _9066,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
066, etc
[trunktollfree] ; Long distance context
accessed through trunk interface
exten =
_901800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkpaypercall] ; Dangerous pay-per call!
exten =
_901900.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkcelular] ; Long distance context
accessed through trunk interface
exten =
_9044ZZ,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten =
_9045ZZ,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;
; Contexts
[international] ; Master context for
international long distance
ignorepat = 9
include = longdistance
include = trunkint
[longdistance] ; Master context for long
distance
ignorepat = 9
include = local
include = trunkld
include = trunktollfree
include = trunkpaypercall
[mercadotecnia]
ignorepat = 9
include = local
[local] ; Master context for local,
toll-free, and iaxtel calls only
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
[record]
exten = s,1,Answer
exten = s,2,Read(RECORD|enter4digits|4)
exten = s,3,Playback(record-instructions)
exten =
s,4,Record(/var/lib/asterisk/sounds/recording/s-${RECORD}|wav)
exten = s,5,Wait(2)
exten =
s,6,Playback(/var/lib/asterisk/sounds/recording/s-${RECORD})
exten = s,7,ResponseTimeout(10)
exten =
s,8,Background(1toaccept2torerecord3torecordanother)
exten = 1,1,Hangup
exten = 2,1,Goto(s,3)
exten = 3,1,Goto(s,2)
[macro-stdexten];
;
; Macro de extensiones estandard:
; ${ARG1} - Extension (Pudimos haver usado
${MACRO_EXTEN} tambien aqui
; ${ARG2} - Aparato(s) a marcar
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface,
20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy,
send to voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything
else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain
[macro-stdexten-viejo] ; Standard extension macro:
; ARG1 es el numero de la extension
; ARG2 es sip al cual voy a marcar
exten = s,1,Dial(${ARG2},20,rt) ; Ring the interface,
20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If
they press #, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If
[asterisk-users] Re: zapata with Tiger3XX compilation error
Ok so I read the Linux 2.6 related README and finally compiled propperly, I thought but at the end I notice that lscpi does report the cards, but I cant modprobe wcfxo nor zaptel and I do have wcfxo.ko in the /lib/modules/2.6.8/extra/ directory, so what gives? This is a Debian Sarge, thanks! # # make clean starts here # make[1]: Entering directory `/usr/src/zaptel-1.4.0/menuselect' rm -f menuselect *.o make[2]: Entering directory `/usr/src/zaptel-1.4.0/menuselect/mxml' /bin/rm -f mxmldoc.o testmxml.o mxml-attr.o mxml-entity.o mxml-file.o mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o mxml-string.o libmxml.a mxmldoc doc/mxml.3 doc/mxmldoc.1 testmxml mxml.xml /bin/rm -f mxmldoc-static libmxml.a /bin/rm -f *.bck *.bak /bin/rm -f config.cache config.log config.status /bin/rm -f -r autom4te*.cache make[2]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect/mxml' make[1]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect' rm -f torisatool makefw tor2fw.h radfw.h rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.4.0 clean make[1]: Entering directory `/usr/src/kernel-source-2.6.8' CLEAN /usr/src/zaptel-1.4.0/wct4xxp CLEAN /usr/src/zaptel-1.4.0/.tmp_versions make[1]: Leaving directory `/usr/src/kernel-source-2.6.8' rm -f xpp/*.ko xpp/*.mod.c xpp/.*o.cmd rm -f xpp/*.o xpp/*.mod.o rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest rm -rf misdn* rm -rf mISDNuser* # # ./configure starts here # checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking how to run the C preprocessor... gcc -E checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for GNU make... make checking for grep... /bin/grep checking for sh... /bin/sh checking for ln... /bin/ln checking for grep that handles long lines and -e... (cached) /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for initscr in -lcurses... yes checking curses.h usability... yes checking curses.h presence... yes checking for curses.h... yes checking for initscr in -lncurses... yes checking for curses.h... (cached) yes checking for newtBell in -lnewt... yes checking newt.h usability... yes checking newt.h presence... yes checking for newt.h... yes checking for usb_init in -lusb... no configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts configure: *** Zaptel build successfully configured *** # # make linux26 starts here # make[1]: Entering directory `/usr/src/zaptel-1.4.0/menuselect' make[2]: Entering directory `/usr/src/zaptel-1.4.0/menuselect' make[3]: Entering directory `/usr/src/zaptel-1.4.0/menuselect/mxml' gcc -O -Wall -c mxml-attr.c gcc -O -Wall -c mxml-entity.c gcc -O -Wall -c mxml-file.c gcc -O -Wall -c mxml-index.c gcc -O -Wall -c mxml-node.c gcc -O -Wall -c mxml-search.c gcc -O -Wall -c mxml-set.c gcc -O -Wall -c mxml-private.c gcc -O -Wall -c mxml-string.c /bin/rm -f libmxml.a /usr/bin/ar crvs libmxml.a mxml-attr.o mxml-entity.o mxml-file.o mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o mxml-string.o a - mxml-attr.o a - mxml-entity.o a - mxml-file.o a - mxml-index.o a - mxml-node.o a - mxml-search.o a - mxml-set.o a - mxml-private.o a - mxml-string.o ranlib libmxml.a make[3]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect/mxml' gcc -Wall -o menuselect.o -g -c -D_GNU_SOURCE menuselect.c gcc -Wall -o menuselect_curses.o -g -c -D_GNU_SOURCE menuselect_curses.c gcc -Wall -o strcompat.o -g -c -D_GNU_SOURCE strcompat.c gcc -g -Wall -o menuselect menuselect.o menuselect_curses.o strcompat.o mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.0/menuselect' gcc gendigits.c -lm -o gendigits ./gendigits tones.h gcc -o makefw makefw.c ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.4.0 modules make[1]: Entering directory `/usr/src/kernel-source-2.6.8' CC [M]
[Asterisk-Users] hi guys, a new newbie here needing help :D
I just installed rpm binaries in a new mandriva and I see a frew error messages with asterisk -vvvcfg, btw I would also like a little guidance to just set up a couple sip phones to start playing with soft phone communication with 3 pcs on the network thanks :) ng '/etc/asterisk/agents.conf': Found [skipping chan_alsa.so] [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Registered custom function IAXPEER May 12 15:50:12 WARNING[6173]: chan_iax2.c:9212 load_module: Unable to open IAX timing interface: No such file or directory == Registered application 'IAX2Provision' == Manager registered action IAXpeers == Manager registered action IAXnetstats == Parsing '/etc/asterisk/iax.conf': Found -- doing lookup for '216.207.245.47' == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == Binding IAX2 to default address 0.0.0.0:4569 == IAX Ready and Listening == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf ng '/etc/asterisk/agents.conf': Found [skipping chan_alsa.so] [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Registered custom function IAXPEER May 12 15:50:12 WARNING[6173]: chan_iax2.c:9212 load_module: Unable to open IAX timing interface: No such file or directory == Registered application 'IAX2Provision' == Manager registered action IAXpeers == Manager registered action IAXnetstats == Parsing '/etc/asterisk/iax.conf': Found -- doing lookup for '216.207.245.47' == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == Binding IAX2 to default address 0.0.0.0:4569 == IAX Ready and Listening == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf [codec_gsm.so] = (GSM/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_gsm: using generic PLC == Registered translator 'gsmtolin' from format gsm to slin, cost 1 May 12 15:50:34 WARNING[6173]: config_old.c:28 ast_load: ast_load is deprecated, use ast_config_load instead! == Parsing '/etc/asterisk/rpt.conf': Found == Registered translator 'lintogsm' from format slin to gsm, cost 3 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
