[Asterisk-Users] connecting to Iconnect here using asterisk
David Lowes
davidl at trafficsales.com
Wed Jun 23 03:50:02 MST 2004
Hi,
I wish to connect several ATA186 Phones to each other, to iconnecthere and
to the PSTN using asterisk.
Please tell the appropriate settings for firewall (ports to open etc.)
sip.conf and extensions.conf(part relevant to iconnect).
Also I would be glad to get a working example of your ATA186 configuration.
I tried searching the mailing lists and several sites but did not find an
answer.
My current configuration:
* Asterisk installed on the Gateway (Bound to internal network and to
Internet) (Not behind NAT).
* Several Cisco ATA186 adapters with SIP firmware (Behind NAT with asterisk
set as their "GkOrProxy").
Current State:
* I manage to place calls to my other internal phones.
* Asterisk does not register at Iconnecthere.
* I am not able to place calls through Iconnecthere.
Thanks,
Configuration files: (commented lines represent options I tried).
n= my Iconnect phone number
p=Iconnect password
u=Iconnect User id
sip.conf
[general]
port = 5060
;bindaddr = 0.0.0.0
;bindaddr = 212.199.7.106
context = from-sip
;callerid=No CallID
;register=nnnnnnnnnnn:pppp at sipauth.deltathree.com/ nnnnnnnnnnn
;register= uuuuuuuu:pppp @natrelay.deltathree.com/phone
;register=uuuuuuuu:pppp at sipauth.deltathree.com/phone
[iconnect]
type=friend
secret=pppp
username=uuuuuuuu
host=sipauth.deltathree.com
;host=natrelay.deltathree.com
;fromdomain=xxx.xxx.xxx.xxx
;dtmfmode=inband
nat=no
;nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G726
;ATA 186 adapter
[dave]
type=friend
username=dave
secret=xxxx
nat=yes
host=dynamic
canreinvite=no
;qualify=200
defaultip=192.168.50.2
;ATA 186 adapter
[french1]
type=friend
username=french1
secret=xxxx
nat=no
host=dynamic
canreinvite=no
qualify=200
defaultip=192.168.50.3
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
ICONNECT1=16463752819
CONSOLE=Console/SIP
;[macro-dialiconnect]
;exten => s,1,SetCallerID(${ICONNECT1})
;exten => s,2,SetCIDName(${MYNAME})
;exten => s,3,Dial(SIP/${ARG1}@iconnect,${ARG2})
;exten => s,4,Playback(new/acnt-or-cir-busy-now)
;exten => s,5,Hangup
;exten => s,104,Playback(new/acnt-or-cir-busy-now)
;exten => s,105,Wait,3
;exten => s,106,Playtones(congestion)
;exten => s,107,Wait,30
;exten => s,108,Playback(new/are-you-still-here)
;exten => s,108,Hangup
[from-sip]
exten => _.,1,NoOp
;exten => _.,1,Macro(record-on,${EXTEN},${CALLERIDNUM})
exten => _.,2,Goto(from-sip-post,${EXTEN},1)
exten => i,1,Hangup
exten => h,1,Hangup
[from-sip-post]
exten => 16463752819,1,Dial(${PHONE1},20,Ttm)
exten => 16463752819,2,Playback(transfer)
exten => 16463752819,3,Macro(dialiconnect,${MYCELLPHONE},20)
exten => 16463752819,4,Voicemail(u${PHONE1VM})
exten => 16463752819,5,Hangup
exten => 16463752819,102,Voicemail(b${PHONE1VM})
exten => 16463752819,103,Hangup
exten => 222,1,Dial(SIP/dave,30,t)
exten => 333,1,Dial(SIP/french1,30,t)
;exten =>
444,1,Dial(SIP/6666xxxxxxxxxxxxxx at uuuuuuuu:pppp at 213.137.73.140,30,t)
exten => 555,1,Dial(SIP/xxxxxxxxxxxxxx at iconnect,30,t)
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => h,1,Hangup
[intern]
exten => _.,1,NoOp
exten => _.,2,Goto(intern-post,${EXTEN},1)
exten => i,1,Hangup
exten => h,1,Hangup
[iconnect-forced]
; Experimental "forced" dialing through iconnect to make calls
; prefixed with "6" go out the iconnect channel. This is to
; test some functionality for inbound connections; feel free
; to comment it out.
;
; Dial out on iconnect and wait for 70 seconds for a connect
;
; If no connection in 70 seconds, jump to fastbusy macro
;
exten => _7XXXXXXXXXXX,1,Macro(dialiconnect,${EXTEN:1},70)
exten => _7XXXXXXXXXX,2,Macro(fastbusy)
[intern-post]
; if someone dials a "9" in front of their number, send out via iconnect
(commercial PSTN gateway)
include => iconnect-forced
David Leon Lowes
System Administrator
davidl at trafficsales.com
The Nation Traffic
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