set {sip_require_timer=false} in your outbound calls or globally <div><br><br><div class="gmail_quote">On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <span dir="ltr"><<a href="mailto:mike.burlingame@me.com" target="_blank">mike.burlingame@me.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">it seems if I get an re-invite from the B-Leg FS add's requires timer and changes the session timer to a high value to the re-invite going to the A-Leg come to find out Acme Packets at our ULC's do not like this and send us back a 420 Bad Extension and the call is disconnected with Reason: Q.850;cause=127;text="INTERWORKING"<br>
<br>
The ULC are stating we need to only have it in our supported and do not pass them a require.<br>
<br>
Re-Invite from B-Leg to FS<br>
------------------------------------------------------------------------<br>
INVITE sip:<a href="tel:16025551212" value="+16025551212">16025551212</a>;phone-context=+1@FS_SERVER:5070 SIP/2.0<br>
Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0<br>
Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1<br>
Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d<br>
Contact: <sip:18475551212@B-LEG_IP:5060;transport=udp><br>
Content-Length: 217<br>
Content-Type: application/sdp<br>
CSeq: 33480808 INVITE<br>
From: <sip:<a href="tel:18475551212" value="+18475551212">18475551212</a>@B-LEG_IP>;tag=100052073<br>
Max-Forwards: 92<br>
Session-Expires: 3600;refresher=uas<br>
Supported: timer<br>
To: <sip:<a href="tel:116025551212" value="+16025551212">116025551212</a>@OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ<br>
User-Agent: FreeSwitch<br>
<br>
v=0<br>
o=- <a href="tel:3308986892" value="+13308986892">3308986892</a> 0 IN IP4 B-LEG_IP<br>
s=Media Server<br>
c=IN IP4 B-LEG_IP<br>
t=0 0<br>
m=audio 51246 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
<br>
Re-Invite from FS to A-Leg<br>
------------------------------------------------------------------------<br>
INVITE sip:16025551212@DID_CARRIER:5060;transport=udp SIP/2.0<br>
Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK<br>
Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on><br>
Max-Forwards: 97<br>
From: <sip:<a href="tel:18475551212" value="+18475551212">18475551212</a>;phone-context=+1@OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB<br>
To: <sip:<a href="tel:16025551212" value="+16025551212">16025551212</a>;phone-context=+1@DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72<br>
Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1<br>
CSeq: 33480811 INVITE<br>
Contact: <sip:18475551212;phone-context=+1@FS_SERVER:5070;transport=udp><br>
User-Agent: FreeSwitch<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY<br>
Require: timer<br>
Supported: timer, precondition, path, replaces<br>
Session-Expires: 64800;refresher=uas<br>
Min-SE: 64800<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 235<br>
X-FS-Support: update_display,send_info<br>
<br>
v=0<br>
o=- <a href="tel:3308979701" value="+13308979701">3308979701</a> 3213293310682935904 IN IP4 B-LEG_IP<br>
s=Media Server<br>
c=IN IP4 B-LEG_IP<br>
t=0 0<br>
m=audio 51246 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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