<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">However and oddly enough Level(3)'s Sonus platform deals with this just fine<div><br></div><div><br><div><div>On Sep 14, 2012, at 5:37 PM, Mike Burlingame <<a href="mailto:mike.burlingame@me.com">mike.burlingame@me.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I agree if it's in the supported line why can it not be required and I am fighting with PaeTec and Global Crossing right now on this issue - They want me to just "Disable" it but I do not feel that is the proper thing to do however to resolve the 420's I might have to give in till I can get them to accept it and not fail on it<div><br></div><div>Both carriers use Acme Packet as the SBC and I have a feeling this is where the hangup is coming in at </div><div><br></div><div><br><div><div>On Sep 14, 2012, at 5:29 PM, Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><p>It's Wong to not accept it. IIRC...there is a better case for why it should be there than for it not being there. It's a rather dumb thing for some of these commercial switches to even care about....I don't want to set a precedence here that I will perpetually change our sip stack and chase after every interop case... Make sure the param is set, I believe it disables it completely going against the rfc but I can't check till Monday. This is part of Sofia not FS itself.</p>
<div class="gmail_quote">On Sep 14, 2012 7:19 PM, "Mike Burlingame" <<a href="mailto:mike.burlingame@me.com">mike.burlingame@me.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">Thanks that seem to help the high session timer value however the require: timer is still present <div><br></div><div><div>Invite from B-Leg --> FreeSwitch</div><div><div>recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:</div>
<div> ------------------------------------------------------------------------</div><div> INVITE <a>sip:18665551212@FreeSwitch:5060</a> SIP/2.0</div><div> Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0</div>
<div> Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1</div><div> Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event</div><div> Max-Forwards: 68</div>
<div> Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac</div><div> From: <<a>sip:18665551212@Carrier_IP</a>>;tag=10.152.0.77+1+63320+3c304bf3</div><div> To: "BURLINGAME MIKE" <<a>sip:13605551212@X.X.X.X</a>>;tag=B688eNFc2yv3e</div>
<div> CSeq: 219015274 INVITE</div><div> Expires: 180</div><div> Contact: <<a>sip:19999218665551212@Carrier_IP:5060;transport=udp</a>></div><div> Organization: MetaSwitch</div><div> Supported: 100rel, resource-priority</div>
<div> Content-Length: 193</div><div> Content-Type: application/sdp</div><div> </div><div> v=0</div><div> o=- 3341834898 3341834898 IN IP4 Carrier_IP</div><div> s=-</div><div> c=IN IP4 Carrier_IP</div><div> t=0 0</div>
<div> m=audio 29864 RTP/AVP 18 0 101</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=ptime:20</div><div> a=silenceSupp:off - - - -</div></div><div><br></div><div><br></div><div><div><br></div><div>Invite from FreeSwitch --> A-Leg</div>
<div>send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:</div><div> ------------------------------------------------------------------------</div><div> INVITE <a>sip:mod_sofia@A-LEG_IP:5070</a> SIP/2.0</div><div> Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg</div>
<div> Route: <<a>sip:ULC;lr=on</a>></div><div> Max-Forwards: 47</div><div> From: <<a>sip:18665551212@ULC</a>>;tag=aXFgDty84N6gK</div><div> To: "BURLINGAME MIKE" <<a>sip:13605551212@A-LEG_IP</a>>;tag=XeZZ4FrX4gXeg</div>
<div> Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5</div><div> CSeq: 33497250 INVITE</div><div> Contact: <<a>sip:18665551212@FreeSwitch:5060;transport=udp</a>></div><div> User-Agent: FreeSwitch</div><div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY</div>
<div> Require: timer</div><div> Supported: timer, precondition, path, replaces</div><div> Session-Expires: 3600;refresher=uas</div><div> Min-SE: 120</div><div> Content-Type: application/sdp</div><div> Content-Disposition: session</div>
<div> Content-Length: 193</div><div> P-hint: outbound</div><div> X-FS-Support: update_display,send_info</div><div> </div><div> v=0</div><div> o=- 3341834898 3341834898 IN IP4 Carrier_IP</div><div> s=-</div><div>
c=IN IP4 Carrier_IP</div><div> t=0 0</div><div> m=audio 29864 RTP/AVP 18 0 101</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=ptime:20</div><div> a=silenceSupp:off - - - -</div></div><div><br></div>
<div><br><div><div>On Sep 14, 2012, at 4:42 PM, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:</div><br><blockquote type="cite">Mike, try comma separating the values instead of using two sets of {}:<br>
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."@".$tp_providerip.";fs_path=sip:OUT_PROXY");<br>
<br>-MC<br><br><div class="gmail_quote">On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <span dir="ltr"><<a href="mailto:mike.burlingame@me.com" target="_blank">mike.burlingame@me.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">I have that set in the bridge command however does not seem to be working as expected.<div><br></div><div>add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$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");<div>
<div><br><div><br></div><div><br></div><div><div><div>On Sep 14, 2012, at 3:16 PM, Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>> wrote:</div>
<br><blockquote type="cite">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" target="_blank">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: <<a>sip:18475551212@B-LEG_IP:5060;transport=udp</a>><br>
Content-Length: 217<br>
Content-Type: application/sdp<br>
CSeq: 33480808 INVITE<br>
From: <sip:<a href="tel:18475551212" value="+18475551212" target="_blank">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" target="_blank">116025551212</a>@OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ<br>
User-Agent: FreeSwitch<br>
<br>
v=0<br>
o=- <a href="tel:3308986892" value="+13308986892" target="_blank">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 <a>sip:16025551212@DID_CARRIER:5060;transport=udp</a> SIP/2.0<br>
Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK<br>
Route: <<a>sip:OpenSIPS_DID_CARRIER_Proxy;lr=on</a>><br>
Max-Forwards: 97<br>
From: <sip:<a href="tel:18475551212" value="+18475551212" target="_blank">18475551212</a>;phone-context=+1@OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB<br>
To: <sip:<a href="tel:16025551212" value="+16025551212" target="_blank">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: <<a>sip:18475551212;phone-context=+1@FS_SERVER:5070;transport=udp</a>><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" target="_blank">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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