[Serusers] nathelper

[Federico R. SANCHEZ Y NAPAL]

Hello: I'm using a NATed UA to terminate a call in the PSTN using a
Cisco 5300. I want to use the RTP relay capability. In the script, I
wrote: 

 

    ...

    force_rtp_proxy();

    rewritehost("200.60.XXX.XXX");

    forward("200.60.XXX.XXX", 5060);

    ...

 

Where 200.60.XXX.XXX is the GTW IP.

 

I get an error in the log: 

    "/usr/sbin/ser[7039]: ERROR: extract_mediaip: no 'c=' in SDP"

 

The SIP Message sent to de GTW has a wrong content-length header, so the
GTW rejects it. Here I copy the message: 

 

INVITE sip:2345674284618 at 200.60.30.24 SIP/2.0

Record-Route: <sip:2345674284618 at 200.32.43.55;ftag=673501644;lr=on>

Via: SIP/2.0/UDP 200.32.43.55;branch=0

Via: SIP/2.0/UDP
200.68.54.46:5060;rport=25127;branch=z9hG4bKFE5D22D6B6C4407884

2396149F64C3DE

From: 1001 <sip:1001 at 200.32.43.55>;tag=673501644

To: <sip:2345674284618 at 200.32.43.55>

Contact: <sip:1001 at 200.68.54.46:25127>

Call-ID: 107D459F-7402-480D-988A-FC87455E1E5D at 192.168.0.15

CSeq: 31862 INVITE

Max-Forwards: 69

Content-Type: application/sdp

voip-gtw04#

User-Agent: X-Lite build 1101

Content-Length: 315296

v=0

o=1001 23178458 23178458 IN IP4 200.68.54.46

s=X-Lite

c=IN IP4 200.32.43.55

t=0 0

m=audio 35052 RTP/AVP 0 8 3 98 97 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:98 iLBC/8000

a=rtpmap:97 speex/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=direction:active

I would appreciate any help about this.

Best Regards,

 

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