I could use some advice on getting my iristel.ca connection to work with Trixbox 2.0. I was beginning to think I was having firewall problems with the SIP setup, so I removed the firewall from the equation. Here is what I'm getting from the Asterisk CLI: xx's are to protect the innocent ;)
-- Executing Dial("SIP/100-b7b09e48", "SIP/sbc1.iristel.net/1416xxxxxxx|300|") in new stack
-- Called sbc1.iristel.net/1416xxxxxxx
-- Got SIP response 400 "Bad Request" back from 209.167.234.109
-- SIP/sbc1.iristel.net-09ff61e0 is circuit-busy
Here is my setup for sip.conf:
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.
[general]
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
externip=38.xxx.xxx.xxx
;localnet=192.168.1.0/24 ; All RFC 1918 addresses are local networks
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
Here is the sip_additional.conf:
; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui
register=1416xxxxxxx:xxxx:xxxxxxxx@sbc1.iristel.net/1416xxxxxxx
[100]
type=friend
secret=test101
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=100@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/100
context=from-internal
canreinvite=no
callerid=device <100>
[200]
type=friend
secret=dingus101
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=200@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/200
context=from-internal
canreinvite=no
callerid=device <200>
[sbc1.iristel.net]
username=xxxxxxxx
type=peer
secret=xxxx
insecure=very
host=sbc1.iristel.net
dtmfmode=rfc2833
callerid=1416xxxxxx
Internally everything works fine, I can check mail, call my extensions. Dailing out over the analogue line works like a charm, its just when I go to dial out with the Iristel SIP connection, I get about a split second of ring, then a all circuits are busy email. Have I set something up wrong? I've used all the info I found at voipinfo.org, what I've managed to scour up here as well as all the appropriate info from the PDF Iristel sent me. I wish I could say their support has been a little more helpful in this situation, but currently they just tell me I have an error on my server.
So anyone with some insight, or who has managed to get their outbound connection working, I'm all ears and open to suggestions. In the correspondance I had with Iristel when I was behind the firewall they said that the SIP INVITE was sending an internal IP, that should be fixed I believe by adding the line externalip= plus I have given the Trixbox an external IP directly to see if that was the issue.
Thanks
-Jon
Member Since:
2007-01-22