Hello everyone,
I am running asterisk (1.4.21.2~dfsg-3) on Debian stable. I have SFLphone on my workstation to make calls through. I have a ipcop gateway that the asterisk server and the workstation are behind. I got a sipgate account to make calls with.
So far I have got SLFphone connected to asterisk and it can make outgoing calls. But I can't get incoming calls. Sipgate always shows my phone as offline and test calls do not show up in sip debug or on my gateway. I configured my NAT setup according to [forums.digium.com/viewtopic.php?t=7854](http://forums.digium.com/viewtopic.php?t=7854)
My ipcop configuration looks like:
UDP DEFAULT IP :5060 - 5082 => 192.168.0.47 : 5060 - 5082 Asterisk --- SIP signaling
UDP DEFAULT IP :10000 - 20000 => 192.168.0.47 :10000 - 20000 Asterisk --- RTP traffic
TCP DEFAULT IP :5060 - 5082 => 192.168.0.47 : 5060 - 5082 Asterisk --- SIP signaling I have a file called sip_nat.conf
NAT=Yes
externip=24.113.159.168
localnet=192.168.0.0/255.255.255.0In my sip.conf I have:
[general]
NAT=Yes
externip=24.113.159.168
localnet=192.168.0.0/255.255.255.0
[1000]
type=friend
context=phones
host=dynamic
secret=its_passtastic
register => my-sip-id:my-sip-password@sipgate/my-sip-id
[sipgate]
type=peer
nat=yes
canreinvite=no
; nat=1
; externip=24.113.159.168
secret=my-sip-password
insecure=invite
username=my-sip-id
defaultuser=my-sip-id
fromuser=my-sip-id
context=sipgate_in
;context=incoming_calls
fromdomain=sipgate.com
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
;#include sip_nat.confIn my extensions.conf I have:
[globals]
[general]
static=yes
; NAT=Yes
; externip=24.113.159.168
; externip=***.***.***.***
; localnet=192.168.0.0/255.255.255.0
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming_calls]
exten => _X.,1.NoOp()
exten => _X.,n,Dial(SIP/1000)
;sexten => my-sip-id,1,Dial(SIP/1000)
[outgoing_calls]
;exten => _X.,1,NoOp()
;exten => _X.,n,Dial(SIP/my_service_provider/${EXTEN})
exten => _X.,1,Set(CALLERID(num)=SIP-ID)
exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)
[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()
[phones]
include => internal
include => outgoing_calls
; Stuff from sipgate
[sipgate_in]
exten => my-sip-id,1,Dial(SIP/1000); <-- instead of extension you should define the corresponding peer
exten => my-sip-id,n,Hangup
; [sipgate_out]
; exten => _X.,1,Set(CALLERID(num)=SIP-ID)
; exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)
; exten => _X.,3,HangupWhenever asterisk starts it gives messages like:
[Sep 14 18:05:46] NOTICE[29237] cdr.c: CDR simple logging enabled.
[Sep 14 18:05:46] NOTICE[29237] loader.c: 160 modules will be loaded.
[Sep 14 18:05:46] WARNING[29237] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Sep 14 18:05:46] NOTICE[29237] res_odbc.c: Adding ENV var: INFORMIXSERVER=my_special_database
[Sep 14 18:05:46] NOTICE[29237] res_odbc.c: Adding ENV var: INFORMIXDIR=/opt/informix
[Sep 14 18:05:46] NOTICE[29237] res_odbc.c: res_odbc loaded.
[Sep 14 18:05:46] NOTICE[29237] pbx_ael.c: Starting AEL load process.
[Sep 14 18:05:46] NOTICE[29237] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.
[Sep 14 18:05:46] NOTICE[29237] pbx_ael.c: File /etc/asterisk/extensions.ael not found; AEL declining load
[Sep 14 18:05:46] WARNING[29237] chan_iax2.c: Unable to open IAX timing interface: No such file or directory
[Sep 14 18:05:46] ERROR[29237] chan_vpb.cc: No Voicetronix cards detected
[Sep 14 18:05:46] NOTICE[29248] chan_sip.c: Peer 'sipgate' is now Reachable. (78ms / 2000ms)With sip debugging on I get messages in the console like:
<--- SIP read from 204.155.28.10:5060 --->
SIP/2.0 200 OK
Record-Route: <sip:204.155.28.10;lr=on;ftag=as41100537>
Via: SIP/2.0/UDP 24.113.159.168:5060;branch=z9hG4bK4eb124d1;rport=5060
From: "asterisk" <sip:asterisk@24.113.159.168>;tag=as41100537
To: <sip:sipgate.com>;tag=ddf9a9f0de4595ec12441b9d8d3ee250.00b0
Call-ID: 2197a5e855066ec040ff95e04c51897e@24.113.159.168
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '2197a5e855066ec040ff95e04c51897e@24.113.159.168' Method: OPTIONS
Reliably Transmitting (NAT) to 204.155.28.10:5060:
OPTIONS sip:sipgate.com SIP/2.0
Via: SIP/2.0/UDP 24.113.159.168:5060;branch=z9hG4bK70c65f61;rport
From: "asterisk" <sip:asterisk@24.113.159.168>;tag=as5c8af6a9
To: <sip:sipgate.com>
Contact: <sip:asterisk@24.113.159.168>
Call-ID: 787c25a77625508a4de5fbbe08a721f3@24.113.159.168
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 15 Sep 2009 04:42:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0Edit: Fixed. My register command was not in the general section.