[asterisk-users] no ring back 180 with SDP

[dave george]

I set it under the sip profile for the box sending calls to asterisk.

[BREKEKE]
type=peer
context=wholesale
host=x.x.x.x
nat=no
canreinvite=no
progressinband=yes
dtmfmode=rfc2833
insecure=port
disallow=all
allow=g729

Thanks,
Dave George

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Friday, June 11, 2010 5:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] no ring back 180 with SDP

Hi!

> I tried no, yes and never in the sip profile for that carrier and it did
> not make a difference.
>
> Look at "progressinband=" in sip.conf.

Just to make sure: Maybe you forgot the SIP RELOAD? 

Are you 100% sure inbound calls arrive with the peer that you set 
progressinband for? Verify this using SIP DEBUG. Sip peer matching can be 
quite confusing in Asterisk.

Philipp


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