[asterisk-users] help with sip configuration for sipgate.de on asterisk 1.4

Tue Jul 17 07:02:46 CDT 2007

hi there,
i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main
computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de,
there i can receive call and make them, i can hear the other end but they can not hear me, this is
only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help
me out here. i have a grandstream 486, which is in my sip.conf the 'gs486 deivce, it is connected
to my debain/asterisk machine, and calls are made through it. I was also wondering how i can
activate the 2nd incoming caller thing, usually i used to hear a clicking in the line when
somebody called me and i could switch between callers, is this possible in asteriks? at least one
of my providers supports this service. I was also wondering if there is a free webinterface for
asterisk, from which i can see incoming calls and also make calls, mainly only sip, i don't mean
trixbox or freepbx. anyhow my main problem is the sipgate.de thing, below are my sip.conf and
extension.conf thx for any help

sip.conf:
[general]
language=en
disable=all
register => user:pass at voip.eutelia.it/number
register => user:pass at budgetphone.nl/number
register => user:pass at sipgate.de/number
register => user:pass at sip.voiparound.com/number
register => user:pass at sip.webcalldirect.com/
register => user:pass at ixcall.net/number
register => user:pass at freedigits.net/number
register => user:pass at sip.messagenet.it:5061/number
context=inbound
bind=0.0.0.0
nat=yes
fromdomain=sshn.net
localnet=10.0.0.0/255.255.255.0
externip=195.xxx.xxx.xxx
srvlookup=yes

[authentication]

[eutelia-out]
;maxexpirey=360000
;defaultexpirey=180000
type=friend
allow=alaw
context=inbound
username=xxxx
secret=xxxxx
fromuser=number
fromdomain=voip.eutelia.it
host=voip.eutelia.it
dtmfmode=inband
realm=voip.eutelia.it
registertimeout=300
canreinvite=no
;registertimeout=9999999999
qualify=200
insecure=very
,allow=alaw
,allow=ulaw
,allow=gsm

[messagenet-out]
auth=user:password at sip.messagenet.it
;auth=md5
realm=sip.messagenet.it
qualify=yes
;maxexpirey=360000
;defaultexpirey=180000
authname=user
authuser=user
canreinvite=no
context=inbound
;dtmf=rfc2833
;dtmfmode=rfc2833
fromdomain=sip.messagenet.it
fromuser=user
host=sip.messagenet.it
port=5061
insecure=very
regexten=user
secret=password
md5secret=xxxxxxxxxxxxxxxxxxxxxxx
type=peer
;user=phone
username=user

[webcalldirect-out]
type=peer
context=inbound
username=user
secret=pass
fromuser=user
fromdomain=sip.webcalldirect.com
host=sip.webcalldirect.com
qualify=no
insecure=very
canreinvite=no
allow=all

[sipgate-out]
type=friend
context=inbound
username=user
secret=pass
host=sipgate.de
fromuser=user
fromdomain=sipgate.de
authuser=pass
qualify=yes
insecure=very
disable=all
;allow=alaw
allow=ulaw
allow=alaw
allow=g729
allow=gsm
allow=slinear

[gs486]
type=friend
context=default
username=user
secret=pass
host=dynamic
,nat=no
canreinvite=yes
dtmfmode=info
call-limit=2
allow=all

[budgetphone-out]
type=friend
context=inbound
secret=pass
username=user
host=budgetphone.nl
fromuser=user
fromdomain=budgetphone.nl
insecure=very
allow=alaw

[freecall-out]
type=peer
username=user
secret=pass
qualify=no
authuser=user
fromuser=user
fromdomain=freecall.com
host=sip.voiparound.com
insecure=very
canreinvite=no
allow=all

[iXcall-out]
type=friend
context=inbound
secret=pass
username=user
host=ixcall.net
fromuser=user
fromdomain=ixcall.net
insecure=very
allow=alaw

[freedigits-out]
allow=alaw
context=inbound
fromdomain=freedigits.net
fromuser=user
host=freedigits.net
insecure=very
secret=pass
type=friend
username=user

extensions.conf:
;!
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Sun Jun 10 16:08:39 2007
;!
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no

[globals]
trunk_1 = SIP/trunk_1
trunk_2 = SIP/trunk_2
trunk_4 = SIP/trunk_4
trunk_6 = IAX2/trunk_6
trunk_7 = IAX2/trunk_7
trunk_8 = SIP/trunk_8

[inbound]
exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)})
;exten => s,n,Answer(SIP/gs486)
exten => s,n,Dial(SIP/gs486,30)
exten => s,n,Hangup()
;exten => number,1,Goto(inbound,s,1)
exten => number,1,Goto(from-eutelia,s,1)
exten => number,1,Goto(from-budgetphone,s,1)
exten => user,1,Goto(from-sipgate,s,1)
exten => user,1,Goto(from-ixcall,s,1)
exten => user,1,Goto(from-freedigits,s,1)
exten => user,1,Goto(inbound,s,1)

[default]
exten => _9.,1,Dial(SIP/${EXTEN:1}@budgetphone-out,30,r)
exten => _8.,1,Dial(SIP/${EXTEN:1}@sipgate-out,30,r)
exten => _7.,1,Dial(SIP/${EXTEN:1}@eutelia-out,30,r)
exten => _6.,1,Dial(SIP/${EXTEN:1}@freecall-out,30,r)
exten => _5.,1,Dial(SIP/${EXTEN:1}@webcalldirect-out,30,r)
exten => _4.,1,Dial(SIP/${EXTEN:1}@iXcall-out,30,r)
exten => _3.,1,Dial(SIP/${EXTEN:1}@freedigits-out,30,r)
exten => _2.,1,Dial(SIP/${EXTEN:1}@messagenet-out,30,r)
exten => 1000,1,Dial(SIP/gs486)
exten => 444,1,Dial(SIP/444 at budgetphone-out,30,r)

[from-budgetphone]
exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)})
exten => s,2,Dial(SIP/gs486,30)

[from-eutelia]
exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)})
exten => s,2,Dial(SIP/gs486,30)

[from-messagenet]
exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)})
exten => s,2,Dial(SIP/gs486,30)

[from-sipgate]
exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)})
exten => s,2,Dial(SIP/gs486,30)

[from-freedigits]
exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)})
exten => s,2,Dial(SIP/gs486,30)

[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
hasbeensetup = N

[numberplan-custom-1]
plancomment = DialPlan1
include = default
exten = _00X!,1,Macro(trunkdial,${trunk_4}/${EXTEN:0})
comment = _00X!,1,webcall,standard

[DID_trunk_1]
include = default

[DID_trunk_2]
include = default

[DID_trunk_3]
include = default

[DID_trunk_4]
include = default

[DID_trunk_5]
include = default

[DID_trunk_6]
include = default

[DID_trunk_7]
include = default

[DID_trunk_8]

include = default

thx for any help 
jody :)

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