Help configuring SIP trunk

Asterisk Asterisk Support
1

I am new to Asterisk, and am still learning. I am currently having problems making out going call over SIP. I have looked around, and I have not been able to find out how to fix this. I have lover looked something in all probability, but I can not see it.

Setup:
I have a SIP line that connects to a MetaSwitch softswitch, that connects to the PSTN.

softphone -> Asterisk Box -> MetaSwitch Softswitch -> PSTN

The problem:
I currently can get incoming calls, but outgoing calls fail with an error “SIP/2.0 403 From: URI not recognized” from the MetaSwitch, and a “SIP/2.0 503 Service Unavailable” returned from the Asterisk box.
I can make a call using the account with just a softphone, leaving out the asterisk box, so I know the sip line works. I can also call other extensions on the asterisk box. I spoke with the Metaswitch maintainer, he said that he could see the call get built to the PSTN, but then it gets torn down maybe a second or two after being built.

Everything is on internal networks, with no NAT between them

Here is my sip.conf file:
[general]
canreinvite=no
useragent = X-Lite release 1103m

register => 8646826933:@10.yy.yy.148

[MetaSwitch]
type=friend
insecure=port,invite
context=incomming
host=10.yy.yy.148
username=864682YYYY
secret=

;;-----------------------------------------------------------------
[1000]
type=friend
host=dynamic
canreinvite=no
deny=0.0.0.0/0.0.0.0
permit=192.zz.zz.0/255.255.254.0
permit=172.xx.xx.0/255.255.252.0
context=myphones

Here is my part of the extensions.conf file:
[general]
static=yes
writeprotect=no

[incoming]
exten => s,1,Log(NOTICE, Incoming call from ${CALLERID(all)})
exten => s,n,Dial(SIP/1000)
exten => s,n,Hangup()
; End of the “incoming” context

[myphones]
;;Dial fixed number
exten => 789,1,Set(CALLERID(num)=864682YYYY)
exten => 789,n,Set(CALLERID(name)=Brian)
exten => 789,n,Dial(SIP/683XXXX@MetaSwitch,20)
exten => 789,n,Congestion

;;Dial and number with 9 as prefix
exten => _9.,1,Set(CALLERID(num)=864682YYYY)
exten => _9.,n,Dial(SIP/${EXTEN:1}@MetaSwitch,60,r)
exten => _9.,n,Congestion

Here is a link to the sip debug trace: backroads.net/home/bslice/SipDebug.txt

Any help, suggestions, or new places to look for example configs would be appreciated very much.

Thanks

2

Are you using 2 NIC on the server? Because I am having a hard time understanding how the 10.yy.yy.148 and 172.xx.xx.0/255.255.252.0 are interconnected.

It’s a longshot, but what happens if you do this: