Help for Asterisk config for calls in and out from PSTN

[visualhello]

Hi all,

Having issues with calls in and out from PSTN to my FPL account.

I had asterisk for my VOIP with FPL and other providers running well in Aug 2010.
Then I had to move and was not able to reconnect until now.
All these time have been able to use x-lite with no problem for all voip providers.

Now in my asterisk box.
I can make and receive calls between FPL users but gives me error for any calls in and out from PSTN .
Error message in asterisk is "process_sdp: Insufficient information in SDP (c=)..."
All my lines for other providers are running fine.

My setting are
trunkname = freephoneline
type = friend
nat = yes
canreinvite = no
hasiax = no
hassip = yes
registeriax = no
registersip = yes
trunkstyle = voip
username = 1XXXXXXXXXX
fromuser = 1XXXXXXXXXX
authuser = 1XXXXXXXXXX
fromdomain = voip.freephoneline.ca
secret = XXXXXXXX
host = voip.freephoneline.ca
dtmfmode = rfc2833
context = DID_freephoneline
insecure = port,invite
disallow = all
allow = g729,ulaw,alaw

Any help would be greatly appreciated.

Thanks

[curriegrad2004]

Getting FPL to work with asterisk is just a matter of registering another SIP trunk. Did you try updating your asterisk software before?

For me, getting FPL to work with a PBX is all good for me, although it's not Asterisk, rather FreeSwitch.

[tbrummell]

They do not allow a useragent=asterisk, have you changed this to something else already? Myself, I use useragent=LinksysPAP2T and it works fine.