[Asterisk-Users] ceptral (swift)

2005-07-20 Thread Mahmoud Badran




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] 







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RE: [Asterisk-Users] modprobe wcfxo fails.

2005-07-20 Thread Mahmoud Badran




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



 



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        Mahmoud Badran



ATSI

Tel: +20 2 607 8917

Fax: +20 2 607 9178

gsm:    +20 105674409

[EMAIL PROTECTED] 







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Re: [Asterisk-Users] asterisk newbie and phones which don't want tocomunicate

2005-07-04 Thread Mahmoud Badran




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

2005-07-02 Thread Mahmoud Badran
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
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RE: [Asterisk-Users] asterisk newbie and phones which don't want tocomunicate

2005-07-01 Thread Mahmoud Badran
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
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RE: [Asterisk-Users] make error for zaptel

2005-07-01 Thread Mahmoud Badran
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 #

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RE: [Asterisk-Users] Got this error after my installation when i doztcfg -vv

2005-07-01 Thread Mahmoud Badran
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'
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RE: [Asterisk-Users] cdr and billing

2005-06-26 Thread Mahmoud Badran




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

2005-06-26 Thread Mahmoud Badran




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] handle wrong extensions in Dialplam

2005-06-26 Thread Mahmoud Badran




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


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[Asterisk-Users] Music on Hold Choppy

2005-06-23 Thread Mahmoud Badran




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






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[Asterisk-Users] H323 trunk with cisco gatekeeper

2005-06-05 Thread Mahmoud Badran




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

2005-05-18 Thread Mahmoud Badran
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??

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[Asterisk-Users] H323 FAX

2005-04-28 Thread Mahmoud Badran
> 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?
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Re: [Asterisk-Users] Cisco 7960s and skinny

2005-04-28 Thread Mahmoud Badran
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 
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> >>
> >>
> >>
> >>
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> >
> 
> 
> -- 
> 
> 
> 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
> 
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[Asterisk-Users] H323 FAX

2005-04-28 Thread Mahmoud Badran
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? 
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