[Asterisk-Users] ceptral (swift)
Hi i installed ceptral and i want to test it with asterisk can u plz tell me if i was wrong here>> ?? exten => 2,1,Answer exten => 2,2,system(/opt/swift/bin/swift "hello world") exten=> 2,3,Hangup() Mahmoud Badran ATSI Tel: +20 2 607 8917 Fax: +20 2 607 9178 gsm: +20 105674409 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] modprobe wcfxo fails.
Sorry but can u remind me which problem?? i've answered many people before with diffrent problems what problem u r having? On Tue, 2005-07-19 at 15:20 -0400, Jerry Rasmussen wrote: Can you help me remember the details. Its been a while since I have touched the system. From: [EMAIL PROTECTED] on behalf of Tim King Sent: Sun 7/17/2005 3:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] modprobe wcfxo fails. I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer & Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mahmoud Badran ATSI Tel: +20 2 607 8917 Fax: +20 2 607 9178 gsm: +20 105674409 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk newbie and phones which don't want tocomunicate
Hiii ; actually you are not allowing any codecs in the sip.conf neither alaw nor ulaw
so try this to all phones in sip.conf or put it in the general context (allow=all)
[2011]
type=friend
username=2011
secret=1945
nat=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=200
allow=all
On Mon, 2005-07-04 at 18:00 +0200, Sistemista WebSolvingJaa wrote:
with some trials configuration,and a couple of hours now i can make a
call from a phone to another phone. typing the code of phone A from
phone B, the ring-tone of phone A rings but neither phone A and phone
B can comunicate as voice (i hope my explaination can be understood by
all of you). so my extension.conf is now like this:
[general]
static=yes
writeprotect=yes
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for demo
CONSOLE=Zap/1
CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
switch => DUNDi/e164
[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch
[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
exten => _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider]
;switch => IAX2/user:[EMAIL PROTECTED]/mycontext
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten => _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[macro-stdexten];
exten => s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${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${ARG1}) ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain
[demo]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,SetVar(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,SetVar(TIMEOUT(response)=10) ; Set Response Timeout
to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct); Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,SetVar(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart); Start with the congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234); Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234); Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call
RE: [Asterisk-Users] play message to callee before connect to incomingcall
try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after From: [EMAIL PROTECTED] on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] play message to callee before connect to incomingcall Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) should be connected to SIP Phone 100 any suggestions on how to implement this in an easy way? Thanks in advance, Roland Zagler ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk newbie and phones which don't want tocomunicate
hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvc to check see them registering... From: [EMAIL PROTECTED] on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk newbie and phones which don't want tocomunicate Hi list, i'm an asterisk newbie and i've to setup a net with an asterisk server and several ip phones linked on the net. i hope my questions are IT ans if you have some link for solving those problems please mail me. i've wrote the sip.conf in this way: [2011] type=friend username=2011 secret=1945 regexten=1234 host=dynamic ;host=192.168.100.242 permit=192.168.100.0/24 context=sip-incoming canreinvite=yes dtmfmode=rfc2833 nat=1 [2012] type=friend username=2012 secret=1945 regexten=1234 host=dynamic ;host=192.168.100.221 permit=192.168.100.0/24 context=sip-incoming canreinvite=yes dtmfmode=rfc2833 nat=1 and the extension.conf if quitelly the same as the original. the phones softwares are setted up correctly, but from a phone i can't call another phone on the net. can somebody suggest me a possible solution? thanks a lot ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] make error for zaptel
maybe zaptel verion incompatability try other newer or stable older versions not sure thats just a hint From: [EMAIL PROTECTED] on behalf of Zoltan Szecsei Sent: Fri 7/1/2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] make error for zaptel Hi, I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the box (in a hope to sort out the uname -r issue mentioned below). I'm using the Asterisk Doc Proj vol 1 to guide me through the initial setup. I have no special HW and intend to use asterisk on an internal network just to get some experience. I have downloaded what I think I need and placed it in /usr/src (see listing below). I run make clean ; make linux26 (what about the usual make with no parameters?) and I get a crash. Note that uname -r returns a *different* version of what the linux is linked to (thanks to YOU??) I have tried make clean ; make (no params) and it still crashes. Can anyone offer me some suggestions? - or do I go first to the SuSE list to sort out the uname -r & usr/src/linux issue? TIA, Zoltan. gl0:/usr/src # ls -la total 499 drwxr-xr-x 17 root root680 2005-06-30 14:46 . drwxr-xr-x 13 root root368 2005-06-30 09:21 .. drwxr-xr-x 3 root root320 2005-05-19 06:53 astcc -rw-r--r-- 1 root root 130943 2005-06-25 19:09 astcc.tar drwxr-xr-x 22 root root 2336 2005-06-23 04:16 asterisk-1.0.8 drwxr-xr-x 7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8 drwxr-xr-x 3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8 drwxr-xr-x 2 root root440 2005-05-23 06:47 btp -rw-r--r-- 1 root root 32975 2005-06-25 19:08 btp.tar drwxr-xr-x 23 root root776 2005-06-12 17:24 dicts drwxr-xr-x 5 root root416 2005-04-01 18:50 gastman -rw-r--r-- 1 root root 332857 2005-06-25 19:08 gastman.tar drwxr-xr-x 25 root root736 2005-06-12 17:29 kernel-modules drwxr-xr-x 2 root root520 2005-06-23 04:11 libpri-1.0.8 lrwxrwxrwx 1 root root 19 2005-06-25 22:01 linux -> linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a drwxr-xr-x 3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj drwxr-xr-x 19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj lrwxrwxrwx 1 root root 23 2005-06-25 22:01 linux-obj -> linux-2.6.11.4-21.7-obj drwxr-xr-x 7 root root168 2005-06-12 17:43 packages drwxr-xr-x 2 root root 2720 2005-07-01 12:43 zaptel-1.0.8 gl0:/usr/src # uname -r 2.6.11.4-20a-smp gl0:/usr/src # cd zaptel-1.0.8/ gl0:/usr/src/zaptel-1.0.8 # make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core gl0:/usr/src/zaptel-1.0.8 # make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c -o makefw ./makefw tormenta2.rbt tor2fw > tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make: *** [linux26] Error 2 gl0:/usr/src/zaptel-1.0.8 # ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update o
RE: [Asterisk-Users] Got this error after my installation when i doztcfg -vv
hello u can see the readme.udev in the zaptel directory that's normally answers ur question From: [EMAIL PROTECTED] on behalf of Ian Bert Tusil Sent: Fri 7/1/2005 9:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got this error after my installation when i doztcfg -vv how can i solve the error on the last part? need help. thnx... Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: Individual Clear channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels configured. Notice: Configuration file is /etc/zaptel.conf line 145: Unable to open master device '/dev/zap/ctl' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr and billing
thanks alot for help but problem is; consider this scenario an internal sip phone calls the IVR which shouldnt be billed then he dial an extension from the ivr that redirects him to outbound line that makes the call have some time counting in the ivr and other time counting during the outbound call so how can i bill him on the outbound only??
On Sun, 2005-06-26 at 15:23 +0300, jurczak wrote:
Well, in your dialplan, in the place where you are calling SIP (internal phones) you should put nocdr in the first priority
So if you would have
Exten => _4.,1,NoCdr
Exten => _4.,2,Dial(SIP/${EXTEN})
Assuming that your SIP begins with 4.
With this you wont have any CDR for your internal calls.
-Original Message-
From: Mahmoud Badran [mailto:[EMAIL PROTECTED]
Sent: Sunday, June 26, 2005 3:17 PM
To: jurczak
Subject: RE: [Asterisk-Users] cdr and billing
come on!!
whats wrong with ya?
On Sun, 2005-06-26 at 15:01 +0300, jurczak wrote:
Why don’t you try nocdr
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mahmoud Badran
Sent: Sunday, June 26, 2005 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] cdr and billing
Hello ;
how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled
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[Asterisk-Users] cdr and billing
Hello ; how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] handle wrong extensions in Dialplam
Hello; i am trying to make a dial plan that can handle any wrong extensions dialled from the local sip phone for example so that if i dialled the right extension it rings but if i dialled wrong or existing extension it redirect him to the Main menu for example? thanks in advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on Hold Choppy
Hello all i am using asterisk 1.07 with mpg123-0.59r but still i get very choppy sounds, any suggestions? extensions.conf --- exten => 444,1,WaitMusicOnHold(120) modules.conf [modules] autoload=yes load => chan_modem.so load => res_musiconhold.so noload => chan_alsa.so noload => chan_oss.so [global] chan_modem.so=yes musiconhold.conf - [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 trunk with cisco gatekeeper
Hi .
AVE!
i am trying to register h323 asterisk to the gatekeeper as i installed
asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323
on fedora core3 on a cisco mcs 7800 server problem is i want the
asterisk to register with gatekeeper endpoint with specific zone name
I installed the asterisk-oh323-0.6.6-pre4 and is working nice as I registered to the cisco gatekeeper zone I wanted the problem is when I try to make acall to pstn number or even a registered sip user in asterisk the h323 tries to make the call but I get the an error in the console (not enough bandwidth) as the log file shows that the (admission request rejected)!!! I'll attach both the h323.conf ,oh323.log and extension.conf files plz help me, is it something from the cisco side or the asterisk configuration file??
; 9 is the ignorepat
; 4800805 is a normal phone line from the pstn
exten => 123,1,Dial(oh323/94800805)
exten => ,1,Dial(sip/xlite1)
[general]
listenAddress=0.0.0.0
;my port is 1719
listenPort=1719
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
;bandwidthLimit=1024
wrapLibTraceLevel=3
libTraceLevel=3
libTraceFile=/var/log/asterisk/oh323.log
[EMAIL PROTECTED]
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=H323
language=en
musiconhold=default
context=default
[register]
context=default
;alias=fax
gwprefix=14002
[codecs]
codec=G711A
frames=20
0:00.003 asterisk-oh323 H323 Created endpoint.
0:00.004 asterisk-oh323 H323 Started listener Listener[ip$*:1719]
0:00.004 asterisk-oh323 H323 Added capability: G.711-ALaw-64k{hw} <1>
0:00.004 asterisk-oh323 H323 Added capability: UserInput/hookflash <2>
0:00.004 asterisk-oh323 H323 Added capability: UserInput/basicString <3>
0:00.004 asterisk-oh323 H323 Added capability: UserInput/dtmf <4>
0:00.004 asterisk-oh323 H323 Added capability: UserInput/RFC2833 <5>
0:00.004 H323 Cleaner H323 Started cleaner thread
0:00.005 H323 Listener:8501488 H323 Awaiting TCP connections on port 1719
0:00.477 GKRegThread:08501af0 H323UDP Binding to interface: :::1
0:00.488 GKRegThread:08501af0 RAS Authenticator H235AnnexD_Procedure1 not active during GRQ SetCapability negotiation
0:00.489 GKRegThread:08501af0 RAS Authenticator CAT not active during GRQ SetCapability negotiation
0:00.489 GKRegThread:08501af0 RAS Authenticator MD5 not active during GRQ SetCapability negotiation
0:00.489 GKRegThread:08501af0 H225 Started gatekeeper discovery of "ip$10.222.2.1"
0:00.489 GKRegThread:08501af0 RAS Gatekeeper discovery on interface: 10.190.29.230:10001
0:00.490 GKRegThread:08501af0 Trans Sending PDU [ip$10.190.29.230:10001/ip$*] : gatekeeperRequest 59939
0:00.490 GkMonitor:85051a8 RAS Background thread started
0:00.496 GKRegThread:08501af0 H225RAS Receiving PDU [ip$10.190.29.230:10001/ip$10.222.2.1:1719] : gatekeeperConfirm 59939
0:00.496 GKRegThread:08501af0 RAS Gatekeeper discovery found ip$10.222.2.1:1719
0:00.496 GKRegThread:08501af0 RAS Set alternate gatekeepers:
[EMAIL PROTECTED]:1719;priority=127
0:00.496 GKRegThread:08501af0 RAS Gatekeeper discovered at: 10.222.2.1:1719 (if=10.190.29.230:10001)
0:00.499 GKRegThread:08501af0 Trans Making request: registrationRequest
0:00.499 GKRegThread:08501af0 Trans Sending PDU [ip$10.190.29.230:10001/ip$10.222.2.1:1719] : registrationRequest 59940
0:00.499 GKRegThread:08501af0 Trans Waiting on response to seqnum=59940 for 3.0 seconds
0:00.501 Transactor:85047b8 Trans Starting listener thread on Transport[remote=ip$10.222.2.1:1719 if=ip$10.190.29.230:10001]
0:00.502 Transactor:85047b8 H225RAS Receiving PDU [ip$10.190.29.230:10001/ip$10.222.2.1:1719] : registrationConfirm 59940
0:00.503 Transactor:85047b8 RAS Registered 622A14600144 with fax
0:00.503 Transactor:85047b8 RAS Set alternate gatekeepers:
[EMAIL PROTECTED]:1719;priority=127
0:00.504 Transactor:85047b8 H225RAS Receiving PDU [ip$10.190.29.230:10001/ip$10.222.2.1:1719] : infoRequest 14070
0:00.508 Transactor:85047b8 Trans Sending PDU [ip$10.190.29.230:10001/ip$10.222.2.1:1719] : infoRequestResponse 14070
0:17.532 ThreadID=0xf4400bb0 H323 Making call to:
0:17.535 ThreadID=0xf4400bb0 H323 Added capability: G.711-ALaw-64k{hw} <1>
0:17.536 ThreadID=0xf4400bb0 H323 Added capability: UserInput/hookflash <2>
0:17.536 ThreadID=0xf4400bb0 H323 Added capability: UserInput/basicString <3>
0:17.536 ThreadID=0xf4400bb0 H323 Added capability: UserInput/dtmf <4>
0:17.536 ThreadID=0xf4400bb0 H323 Added capability: UserInput/RFC2833 <5>
0:17.536 ThreadID=0xf4400bb0 H323 Found capability: G.711-ALaw-64k{hw} <1>
0:17.536 ThreadID=0xf4400bb0 H323 Found capability: UserInput/hoo
[Asterisk-Users] Asterisk H323 Trunk Zone
AVE! i am trying to register h323 asterisk to the gatekeeper as i installed asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323 on fedora core3 on a cisco mcs 7800 server problem is i want the asterisk to register with gatekeeper endpoint with specific zone name and type... i searched the web, mail list but there weren't any helpful ones could anyone plz tell me how to specify the zone name and type?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 FAX
> my problem is i need to configure H323/Fax in asterisk to catch H323/Fax > from the gateway and route it as t38/fax to another pbx server i > installed on windows. > > how can i configure, route and convert the faxes? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960s and skinny
i have the cisco 7940 7960 phones and both i can use sip with asterisk? how can i help? On Thu, 2005-04-28 at 10:48 +0100, Derek Conniffe wrote: > Hi Paul, > > I had the same situation - I had a 7940 with only the callmanager > firmware but would have much rathered SIP. You need to have a support > contrace with Cisco to be able to download the firmware from their website. > > Thankfully the support contract only costs about $9 for the year - I was > able to buy the contract from CDW ([EMAIL PROTECTED]) - the product code > for the contract is CON-SNT-CP7940 (replace the 7940 at the end with > 7960 for your phone). > > I'm in Ireland so it seems there is no problem purchasing > internationally either. > > Derek > > Paul wrote: > > >Do you still have that image for the 7960? I bought a 7940 on ebay and it > >doesn't have the SIP firmware. I can't find it anywhere but Cisco's website > >and they require that I have an account with them. Did you happen to save > >that binary file? > > > >Paul > > > >-Original Message- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton > >Sent: Tuesday, April 12, 2005 16:38 > >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] Cisco 7960s and skinny > > > >Simon: > > > >I have had Skinny going on a 7960 (which I then reimaged to SIP). I > >currently run a 7910 on Skinny (using chan_sccp) and use the > >aforementioned 7960 simultaneously. > > > >Since you mentioned that you will have 50 phones, I assume you are > >using them in a business setting. I would *highly* recommend using > >SIP, as I have found that the skinny driver is not as reliable as it > >could be (not criticizing Jan or Julien at all, here). > > > >Reimaging the 50 of them should only take a while (depending on what > >version of CCM they have at the moment). I reimaged 12 phones once for > >a business and it took less than 30 minutes after I got it going > >(toying with the phones to get them to take the image, exactly how the > >config files were to be set up, etc...). > > > >I imagine you could easily get the whole thing done in less than a day > >(reimaging and config files), then figure out your dialplan. > > > >Then there is the whole issue of writing the config files...but you'd > >have to do those with Skinny, anyhow. I think with SIP you'll have > >much better reliability. > > > >-Andy > >FWD: 428725 > > > >On Apr 12, 2005 12:48 PM, Morris, Simon <[EMAIL PROTECTED]> wrote: > > > > > >> > >> > >>Hello, > >> > >> Does anyone else have * running with Cisco 7960 phones and skinny? > >> > >> All the advise I am reading so far is telling me to load the SIP image on > >>the phone but I'd like to know what I'm going to lose by persisting with > >>skinny > >> > >> (Not reimaging 50 phones is one benefit amongst others of skinny) > >> > >> Thanks for any comparisons you can provide > >> > >> Rgds > >> > >> ~sm > >>___ > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> > >> > >___ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >___ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > -- > > > Derek Conniffe > Rivertower Ltd > DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 > Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 > Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 > Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 > Email: [EMAIL PROTECTED] > Web: www.rivertowerhosting.com > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 FAX
hello i successfully installed asterisk on fedora core 3 and all what's in the check list plus the ACTOS gui and asterisk manager but i used actos to configure my cisco ip phones and dial/receive calls through sip. my problem is i need to configure H323/Fax in asterisk to catch H323/Fax from the gateway and route it as t38/fax to another pbx server i installed on windows. how can i configure, route and convert the faxes? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
