<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I have that set in the bridge command however does not seem to be working as expected.<div><br></div><div>add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");<br><div><br></div><div><br></div><div><div><div>On Sep 14, 2012, at 3:16 PM, Anthony Minessale &lt;<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">set {sip_require_timer=false} in your outbound calls or globally&nbsp;<div><br><br><div class="gmail_quote">On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <span dir="ltr">&lt;<a href="mailto:mike.burlingame@me.com" target="_blank">mike.burlingame@me.com</a>&gt;</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>
&nbsp; &nbsp;------------------------------------------------------------------------<br>
&nbsp; &nbsp;INVITE sip:<a href="tel:16025551212" value="+16025551212">16025551212</a>;phone-context=+1@FS_SERVER:5070 SIP/2.0<br>
&nbsp; &nbsp;Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0<br>
&nbsp; &nbsp;Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1<br>
&nbsp; &nbsp;Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d<br>
&nbsp; &nbsp;Contact: &lt;<a href="sip:18475551212@B-LEG_IP:5060;transport=udp">sip:18475551212@B-LEG_IP:5060;transport=udp</a>&gt;<br>
&nbsp; &nbsp;Content-Length: 217<br>
&nbsp; &nbsp;Content-Type: application/sdp<br>
&nbsp; &nbsp;CSeq: 33480808 INVITE<br>
&nbsp; &nbsp;From: &lt;sip:<a href="tel:18475551212" value="+18475551212">18475551212</a>@B-LEG_IP&gt;;tag=100052073<br>
&nbsp; &nbsp;Max-Forwards: 92<br>
&nbsp; &nbsp;Session-Expires: 3600;refresher=uas<br>
&nbsp; &nbsp;Supported: timer<br>
&nbsp; &nbsp;To: &lt;sip:<a href="tel:116025551212" value="+16025551212">116025551212</a>@OpenSIPs_DID_Proxy&gt;;tag=Dj92X5t8065FQ<br>
&nbsp; &nbsp;User-Agent: FreeSwitch<br>
<br>
&nbsp; &nbsp;v=0<br>
&nbsp; &nbsp;o=- <a href="tel:3308986892" value="+13308986892">3308986892</a> 0 IN IP4 B-LEG_IP<br>
&nbsp; &nbsp;s=Media Server<br>
&nbsp; &nbsp;c=IN IP4 B-LEG_IP<br>
&nbsp; &nbsp;t=0 0<br>
&nbsp; &nbsp;m=audio 51246 RTP/AVP 0 101<br>
&nbsp; &nbsp;a=rtpmap:0 PCMU/8000<br>
&nbsp; &nbsp;a=rtpmap:101 telephone-event/8000<br>
&nbsp; &nbsp;a=fmtp:101 0-15<br>
&nbsp; &nbsp;a=ptime:20<br>
&nbsp; &nbsp;------------------------------------------------------------------------<br>
<br>
Re-Invite from FS to A-Leg<br>
&nbsp; ------------------------------------------------------------------------<br>
&nbsp; &nbsp;INVITE <a href="sip:16025551212@DID_CARRIER:5060;transport=udp">sip:16025551212@DID_CARRIER:5060;transport=udp</a> SIP/2.0<br>
&nbsp; &nbsp;Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK<br>
&nbsp; &nbsp;Route: &lt;<a href="sip:OpenSIPS_DID_CARRIER_Proxy;lr=on">sip:OpenSIPS_DID_CARRIER_Proxy;lr=on</a>&gt;<br>
&nbsp; &nbsp;Max-Forwards: 97<br>
&nbsp; &nbsp;From: &lt;sip:<a href="tel:18475551212" value="+18475551212">18475551212</a>;phone-context=+1@OpenSIPS_DID_CARRIER_Proxy:5060;user=phone&gt;;tag=c9Favaa53XFXB<br>
&nbsp; &nbsp;To: &lt;sip:<a href="tel:16025551212" value="+16025551212">16025551212</a>;phone-context=+1@DID_CARRIER:5060;user=phone&gt;;tag=SDd626401-gK095bbb72<br>
&nbsp; &nbsp;Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1<br>
&nbsp; &nbsp;CSeq: 33480811 INVITE<br>
&nbsp; &nbsp;Contact: &lt;<a href="sip:18475551212;phone-context=+1@FS_SERVER:5070;transport=udp">sip:18475551212;phone-context=+1@FS_SERVER:5070;transport=udp</a>&gt;<br>
&nbsp; &nbsp;User-Agent: FreeSwitch<br>
&nbsp; &nbsp;Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY<br>
&nbsp; &nbsp;Require: timer<br>
&nbsp; &nbsp;Supported: timer, precondition, path, replaces<br>
&nbsp; &nbsp;Session-Expires: 64800;refresher=uas<br>
&nbsp; &nbsp;Min-SE: 64800<br>
&nbsp; &nbsp;Content-Type: application/sdp<br>
&nbsp; &nbsp;Content-Disposition: session<br>
&nbsp; &nbsp;Content-Length: 235<br>
&nbsp; &nbsp;X-FS-Support: update_display,send_info<br>
<br>
&nbsp; &nbsp;v=0<br>
&nbsp; &nbsp;o=- <a href="tel:3308979701" value="+13308979701">3308979701</a> 3213293310682935904 IN IP4 B-LEG_IP<br>
&nbsp; &nbsp;s=Media Server<br>
&nbsp; &nbsp;c=IN IP4 B-LEG_IP<br>
&nbsp; &nbsp;t=0 0<br>
&nbsp; &nbsp;m=audio 51246 RTP/AVP 0 101<br>
&nbsp; &nbsp;a=rtpmap:0 PCMU/8000<br>
&nbsp; &nbsp;a=rtpmap:101 telephone-event/8000<br>
&nbsp; &nbsp;a=fmtp:101 0-15<br>
&nbsp; &nbsp;a=ptime:20<br>
&nbsp; &nbsp;------------------------------------------------------------------------<br>
<br>
<br>
_________________________________________________________________________<br>
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Official FreeSWITCH Sites<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>
Twitter: <a href="http://twitter.com/FreeSWITCH_wire">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net/">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:+19193869900<br>
</div>
_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></div></body></html>