asterisk users - Sep 2005 - Early Media in 180 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 180 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.
.

Hi :)

Am Montag, 26. September 2005 19:48 schrieb Ronald
Voermans:
> Hello,
>
> 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?
>
Well, I am not that expert but AFAIK your PSTN gateway should send a 183 
(Session progress) than a simple 180.
Do you use Dial(SIP/blah|30|m(moh_class)) to start early media?

Regards,
Hauke

If guess I figured it out already.

I made some changes in chan_sip.c (when ringing was received, it didn't
check for SDP), and recompiled.

It's working now! 

Ronald
-------------------------------------

-----Oorspronkelijk bericht-----
Van: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Namens Hauke Zuehl
Verzonden: dinsdag 27 september 2005 10:02
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Early Media in 180 Ringing

Hi :)

Am Montag, 26. September 2005 19:48 schrieb Ronald
Voermans:
> Hello,
>
> 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?
>
Well, I am not that expert but AFAIK your PSTN gateway should send a 183
(Session progress) than a simple 180.
Do you use Dial(SIP/blah|30|m(moh_class)) to start early media?

Regards,
Hauke
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