[SR-Users] Trouble with max-forwards

[Mike Dunton]

Readers,

I am having issues with passing the max-forward header to my freeswitch
service.

Here are the sip invites

Here is a call coming from a kazoo box though my carrier Into kamailio.
Kamailio shows it has Max-Forward after the sanitize check, but when it
reaches freeswitch no Max-Forward Header.

*KAMAILIO*

kamailio[2687]: ERROR: <script>: After Sanitize - [INVITE
sip:+1850764####@aio07-bandwidth.tresta-aio.com:5060
<http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0
Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE
Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.2791522.0
Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.f8621604.0
Via: SIP/2.0/UDP 67.231.9.###:5060;branch=z9hG4bK04B40f39b0e10295cd4
From:  <sip:+1NUMBERSCRUBED at 67.231.9.59;isup-oli=0>;tag=gK043cdaa8
To: <sip:+1850764####@67.231.8.85 <sip%3A%2B18507646071 at 67.231.8.85>>
Call-ID: 537168748_70701748 at 67.231.9.# <537168748_70701748 at 67.231.9.59>##
CSeq: 1289511105 INVITE
*Max-Forwards: 49*
Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED at 67.231.9.###:5060
<http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
X-FS-Support: update_display,send_info
Supported: precondition
Content-Length:   359
Content-Type: application/sdp
Remote-Party-ID:  <sip:+1NUMBERSCRUBED at 67.231.9.###:5060
<http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no

v=0
o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59
s=SIP Media Capabilities
c=IN IP4 67.231.9.72
t=0 0
m=audio 23918 RTP/AVP 9 0 18 96 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:30



*FREESWITCH*
recv 1479 bytes from udp/[10.1.13.123]:5060 at 18:31:26.312854:
   ------------------------------------------------------------------------
   INVITE sip:+1850764###@aio07-bandwidth.tresta-aio.com:5060
<http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0
   Record-Route: <sip:10.1.13.###;lr=on;ftag=gK043cdaa8>
   Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
   Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
   Accept: application/sdp
   Allow: INVITE,ACK,CANCEL,BYE
   Via: SIP/2.0/UDP 10.1.13.123;branch=z9hG4bKc892.1b308e66.0
   Via: SIP/2.0/UDP 67.231.8.195;branch=z9hG4bKc892.2791522.0
   Via: SIP/2.0/UDP 67.231.8.85;branch=z9hG4bKc892.f8621604.0
   Via: SIP/2.0/UDP 67.231.9.59:5060;branch=z9hG4bK04B40f39b0e10295cd4
   From:  <sip:+1NUMBERSCRUBED at 67.231.9.#
<sip%3A%2B1NUMBERSCRUBED at 67.231.9.59>##;isup-oli=0>;tag=gK043cdaa8
   To: <sip:+1850764####@67.231.8.# <sip%3A%2B18507646071 at 67.231.8.85>##>
   Call-ID: 537168748_70701748 at 67.231.9.# <537168748_70701748 at 67.231.9.59>##
   CSeq: 1289511105 INVITE
   MaxForwards would go here?
   Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED at 67.231.9.###:5060
<http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>
   Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
   Supported: precondition
   Content-Length:   359
   Content-Type: application/sdp
   Remote-Party-ID:  <sip:+1NUMBERSCRUBED at 67.231.9.###:5060
<http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no
   X-AUTH-IP: 67.231.8.195

   v=0
   o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59
   s=SIP Media Capabilities
   c=IN IP4 67.231.9.72
   t=0 0
   m=audio 23918 RTP/AVP 9 0 18 96 101
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:96 iLBC/8000
   a=fmtp:96 mode=30
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sendrecv
   a=maxptime:30

Here is my kamailio default.cfg http://pastebin.com/vxdFe8n0

Can anyone point me in the right direction, and tell me why freeswitch
isn't being passed this header? Both kamailio and freeswitch are on the
same box in this example.

Thanks for your help.
Mike
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