login | register
Wed 01 of Aug, 2007 [09:24 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.34s
  • Memory usage: 2.21MB
  • Database queries: 31
  • GZIP: Disabled
  • Server load: 2.93

Asterisk Lucent iMerge Configuration

How to configure Asterisk with a Lucent iMerge


Configure asterisk to use the asterisk-oh323 driver


I have not been able to get the default NuFone driver packaged with Asterisk to work with the iMerge. The inAccess H323 driver is available at http://www.inaccessnetworks.com/projects/asterisk-oh323 I used version 0.6.5 in conjunction with asterisk-1.0.1. Follow the installation and configuration instructions included with the source download.

Once the inAccess driver is installed, remove the NuFone driver by moving chan_h323.so out of /usr/lib/asterisk/modules and move h323.conf from /etc/asterisk to another location. You should have chan_oh323.so and oh323.conf in their place, respectively. Make sure your library path is set correctly as stated in the instructions.

Update:

The iMerge does not understand the H.225 InformationRequestReponses returned by the oh323 driver resulting in dropped calls after 40-90 seconds. This may be an incompatibily between v2 (iMerge) and v4 (oh323) of the H.225 spec. Here's the fix:

Modify openh323/src/gkclient.cxx in the openh323-Janus_patch4 source.

In ::OnReceiveInfoRequest, comment out lines 1889 to 1910

from

if (irq.m_callReferenceValue == 0) {

to

} // end of if-else block

This effectively prevents the inclusion of the perCallInfo fields into the IRR, which the iMerge apparently does not like. You will get a compiler warning about 'H225_InfoRequestResponse &irr' not getting used. This can be ignored.

Recompile the asterisk-oh323 driver with 'make clean && make'. The asterisk-oh323 can be compiled with statically linked oh323 and pwlib libraries (the default since 0.6.4), which decreases Asterisk load time and makes the module a lot easier to maintain.


Configuration


Allow the following ports through your firewall:

1718-1721 tcp/udp : h.323
10000:20000 udp : rtp (range specified in oh323.conf)


-- oh323.conf --

Make sure fastStart=yes and that G711U is the default codec. Outgoing calls from Asterisk did not work corectly with G711A. I have not tested any of the other supported codecs.


;
; Configuration file of OpenH323 channel driver
;

;-----------------------------------------
; General configuration options
; (ports, jitter, GK, ...)
;-----------------------------------------
general
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Configure TCP port range to be used by H.323
;
tcpStart=10000
tcpEnd=20000
;
; Configure UDP port range to be used by H.323
Note
The port range used by RTP are configured from

; "rtp.conf"
;
udpStart=10000
udpEnd=20000
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=no
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Enable in-band-DTMF detection.
(Note
Netmeeting uses in-band DTMFs)

;
inBandDTMF=yes
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
Valid values for this option are

; lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
;
; Disable gatekeeper or specify a gatekeeper.
Valid values for this option are

; DISABLE,
; DISCOVER,
; <gatekeeper's DNS name>,
; <gatekeeper's ip>,
GKID
<gatekeeper's id>

;
gatekeeper=<iMerge GK ip>
;
;
; Set the gatekeeper password
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout
;
gatekeeperTTL=600
;
; Set the mode for sending user-input
Valid values for this option are

; Q931 - Q.931 Keypad Information Element
; STRING - H.245 string
; TONE - H.245 tone
; RFC2833 - RFC2833
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Set the default context of H.323 calls.
;
context=local

;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
register
;
; Aliases/prefixes associated with the default context
; defined in section general.
;
context=local
alias=<registered number>
;
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

;-----------------------------------------
; Specify and configure CODEC related
; options
;-----------------------------------------
codecs
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
Valid values for the "codec" option are

; G711U - G.711 u-Law
; G711A - G.711 A-Law
; G7231 - G.723.1(6.3k)
; G72316K3 - G.723.1(6.3k)
; G72315K3 - G.723.1(5.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G726 - G.726(32k)
; G72616K - G.726(16k)
; G72624K - G.726(24k)
; G72632K - G.726(32k)
; G72640K - G.726(40k)
; G728 - G.728
; G729 - G.729
; G729A - G.729A
; G729B - G.729B
; G729AB - G.729AB
; GSM0610 - GSM 0610
; MSGSM - Microsoft GSM Audio Capability
; LPC10 - LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;
;codec=G711A
;frames=20
codec=G711U
frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
;codec=G729
;frames=2



-- extensions.conf --

Outgoing calls: OH323/<exten>



general
static=yes
writeprotect=no

globals
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)

IAXTRUNK1=IAX2/uname:passwd@vpconnect-t01
IAXTRUNK2=IAX2/uname:passwd@vpconnect-t02

PHONE1=SIP/test-laptop

iaxtel700
exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)

iaxprovider

trunkint
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion

trunkld
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion

trunklocal
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion

trunktollfree
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion

international
ignorepat => 9
include => longdistance
include => trunkint

longdistance
ignorepat => 9
include => local
include => trunkld

local
exten => 21,1,Dial(${PHONE1},30,t)
exten => 21,2,Voicemail(u21)
exten => 21,3,Hangup
exten => 21,102,Voicemail(u21)
exten => 21,103,Hangup

exten => 22,1,Dial(${PHONE2}&${PHONE5},30,t)
exten => 22,2,Voicemail(22)
exten => 22,3,Hangup
exten => 22,102,Voicemail(u22)
exten => 22,103,Hangup

exten => 23,1,Dial(${PHONE3},30,t)
exten => 23,2,Voicemail(u23)
exten => 23,3,Hangup
exten => 23,102,Voicemail(u23)
exten => 23,103,Hangup

exten => 24,1,Dial(${PHONE5},30,t)
exten => 24,2,Voicemail(u22)
exten => 24,3,Hangup
exten => 24,102,Voicemail(u22)
exten => 24,103,Hangup

exten => 99,1,Dial(${PHONE4},30,t)
exten => 99,2,Voicemail(u99)
exten => 99,3,Hangup
exten => 99,102,Voicemail(u99)
exten => 99,103,Hangup


;Allow outgoing H323 calls
;No need to specify GK ip if GK is set in oh323.conf
exten => _XXXXXXXXXX,1,Wait,2
exten => _XXXXXXXXXX,2,Dial(OH323/${EXTEN});

;Accept incoming H323 calls
exten => <H323 alias>,1,Wait,1
exten => <H323 alias>,2,Answer
exten => <H323 alias>,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => <H323 alias>,4,ResponseTimeout,10
exten => <H323 alias>,5,Background(demo-thanks)
exten => <H323 alias>,6,Hangup

ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
include => outbound


macro-stdexten;
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they press #, return to start
exten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s,103,Goto(default,s,1) ; If they press #, return to start

voicepulse-in
include => default
include => local

exten => <vp number>,1,Wait,1 ; Wait a second, just for fun
exten => <vp number>,2,Answer ; Answer the line
exten => <vp number>,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => <vp number>,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => <vp number>,5,BackGround(demo-thanks)


default
exten => 85,1,VoicemailMain
exten => 85,2,Hangup

Created by kissel, Last modification by kissel on Thu 10 of Feb, 2005 [16:48 UTC]

Comments Filter

Re: Multiple outbound calls

by gray on Friday 25 of February, 2005 [14:30:10 UTC]
Yup, I see that too. We are working it, but time is short right now.
Edit

Multiple outbound calls

by Anonymous on Wednesday 16 of February, 2005 [14:24:19 UTC]
(:sad:)
I've run into a problem with multiple outbound calls.
If two people dial out the first caller is dropped and the second caller is connected to whoever the first person was talking to.

Does anyone have a work around for this?

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver