SIP Phone to Phone via Gateway: Called Party is Busy

Header:

c=IN IP4 192.168.26.10

Header:

t=0 0

Header:

m=audio 26268 RTP/AVP 0 101

Header:

a=rtpmap:0 pcmu/8000

Header:

a=rtpmap:101 telephone-event/8000

Header:

a=fmtp:101 0-11

-----------------------------------------------------------------

 

SIP Headers

-----------------------------------------------------------------

sip-res:

SIP/2.0 302 Moved Temporarily [192.168.26.200:5060->192.168.26.180:5060]

Header:

Via: SIP/2.0/UDP 192.168.26.180:5060;branch=1

Header:

Via: SIP/2.0/UDP 192.168.26.10:5060

Header:

From: <sip:6711@192.168.26.10:5060>

Header:

To: <sip:93831069@192.168.26.180:5060>

Header:

Call-ID: c2943000-3e262-e4dc-2e323931@192.168.26.10

Header:

CSeq: 100 INVITE

Header:

Contact: <sip:93831069@192.168.26.110:5060;user=phone>

Header:

Content-Length: 0

-----------------------------------------------------------------

 

SIP Headers

-----------------------------------------------------------------

sip-req:

ACK sip:93831069@192.168.26.200:5060;user=phone SIP/2.0 [192.168.26.180:5060-

>192.168.26.200:5060]

Header:

Via: SIP/2.0/UDP 192.168.26.180:5060;branch=1

Header:

From: <sip:6711@192.168.26.10:5060>

Header:

To: <sip:93831069@192.168.26.180:5060>

Header:

Call-ID: c2943000-3e262-e4dc-2e323931@192.168.26.10

Header:

CSeq: 100 ACK

Header:

Content-Length: 0

-----------------------------------------------------------------

 

SIP Headers

-----------------------------------------------------------------

sip-req:

INVITE sip:93831069@192.168.26.110:5060;user=phone SIP/2.0

[192.168.26.180:5060->192.168.26.110:5060]

Header:

Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2

Header:

Via: SIP/2.0/UDP 192.168.26.10:5060

Header:

From: <sip:6711@192.168.26.10:5060>

Header:

To: <sip:93831069@192.168.26.180:5060>

Header:

Call-ID: c2943000-3e262-e4dc-2e323931@192.168.26.10

Header:

CSeq: 100 INVITE

Header:

Proxy-Authorization: Basic 123

Header:

Expires: 180

Header:

Record-Route:

<sip:93831069@192.168.26.180:5060;maddr=192.168.26.180>

Header:

Contact: <sip:6711@192.168.26.10:5060>

Header:

Content-Type: application/sdp

Header:

Content-Length: 221

-----------------------------------------------------------------

 

SDP Headers

-----------------------------------------------------------------

Header:

v=0

Header:

o=CiscoSystemsSIP-IPPhone-UserAgent 16302 19778 IN IP4

192.168.26.10

Header:

s=SIP Call

Header:

c=IN IP4 192.168.26.10

Header:

t=0 0

Header:

m=audio 26268 RTP/AVP 0 101

Header:

a=rtpmap:0 pcmu/8000

Header:

a=rtpmap:101 telephone-event/8000

Header:

a=fmtp:101 0-11

-----------------------------------------------------------------

 

SIP Headers

-----------------------------------------------------------------

sip-res:

SIP/2.0 100 Trying [192.168.26.110:5060->192.168.26.180:5060]

Header:

Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2

Header:

Via: SIP/2.0/UDP 192.168.26.10:5060

Header:

From: <sip:6711@192.168.26.10:5060>

Header:

To: <sip:93831069@192.168.26.180:5060>

Header:

Call-ID: c2943000-3e262-e4dc-2e323931@192.168.26.10

Header:

CSeq: 100 INVITE

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Cisco Systems 1.3.0 manual SIP Phone to Phone via Gateway Called Party is Busy