Hello Ronald,
A 180 Ringing is something that should not have SDP because it's out of band
signaling of the exact status of the call, ringing. The PSTN Gateway should
return a 183 Session Progress if it wants to deliver inband audio progress.
Their SIP implementation doesn't look the best either... so to get it to
work you'd either have to hack Asterisk, or get the manufacturer of the PSTN
gateway to fix their stuff.
Joshua Colp
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ronald
Voermans
Sent: Monday, September 26, 2005 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Early Media in 100 Ringing
Hello,
I have a problem with the following: When I dial a PSTN number from a
UAC, the call is made through a SIP Trunk (which has a connection to the
PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but
Asterisk forwards the 100 Ringing WITHOUT SDP:
As you can see below, the SIP message from 10.254.254.1 (the PSTN
Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content.
How can this be solved?
U 10.254.254.1:5060 -> 192.168.0.173:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35.
Record-Route: <sip:0161801019@10.166.38.108:5060>.
Record-Route: <sip:0161888874@10.254.254.1:5060;lr;nat=yes>.
From: "0161801019"
<sip:0161801019@192.168.0.173>;tag=as02de1b95.
To: <sip:0161888874@10.254.254.1>;tag=00-04094-52dbe3bc-6cf68a723.
Call-ID: 71f7297e0e6cc0625bbae5be00f8a2cc@192.168.0.173.
CSeq: 102 INVITE.
Contact: <sip:212.241.48.70:5060>.
server: Cirpack/v4.38f (gw_sip).
Allow: UPDATE, REFER.
Content-Type: application/sdp.
Content-Length: 253.
.
v=0.
o=cp10 112775383044 112775383045 IN IP4 10.166.38.109.
s=SIP Call.
c=IN IP4 10.254.254.1.
t=0 0.
m=audio 35058 RTP/AVP 18 101.
b=AS:64.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000/1.
a=fmtp:101 0-15.
a=ptime:20.
#
U 192.168.0.173:5060 -> 192.168.1.103:5062
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265.
From: "411" <sip:411@192.168.0.173>;tag=f93ee2f65c6906cb.
To: <sip:0161888874@192.168.0.173>;tag=as675f246d.
Call-ID: 56dc51e7f5084d4b@192.168.1.103.
CSeq: 60590 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:0161888874@192.168.0.173>.
Content-Length: 0.
.
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