[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



 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040623/71e80c9b/attachment.htm


More information about the asterisk-users mailing list