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.

On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame@me.com> wrote:
Thanks that seem to help the high session timer value however the require: timer is still present 

Invite from B-Leg --> FreeSwitch
recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
   ------------------------------------------------------------------------
   Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
   Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
   Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
   Max-Forwards: 68
   Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
   From: <sip:18665551212@Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
   To: "BURLINGAME MIKE" <sip:13605551212@X.X.X.X>;tag=B688eNFc2yv3e
   CSeq: 219015274 INVITE
   Expires: 180
   Organization: MetaSwitch
   Supported: 100rel, resource-priority
   Content-Length: 193
   Content-Type: application/sdp
   
   v=0
   o=- 3341834898 3341834898 IN IP4 Carrier_IP
   s=-
   c=IN IP4 Carrier_IP
   t=0 0
   m=audio 29864 RTP/AVP 18 0 101
   a=rtpmap:101 telephone-event/8000
   a=ptime:20
   a=silenceSupp:off - - - -



Invite from FreeSwitch --> A-Leg
send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
   ------------------------------------------------------------------------
   INVITE sip:mod_sofia@A-LEG_IP:5070 SIP/2.0
   Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
   Route: <sip:ULC;lr=on>
   Max-Forwards: 47
   From: <sip:18665551212@ULC>;tag=aXFgDty84N6gK
   To: "BURLINGAME MIKE" <sip:13605551212@A-LEG_IP>;tag=XeZZ4FrX4gXeg
   Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
   CSeq: 33497250 INVITE
   User-Agent: FreeSwitch
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
   Require: timer
   Supported: timer, precondition, path, replaces
   Session-Expires: 3600;refresher=uas
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 193
   P-hint: outbound
   X-FS-Support: update_display,send_info
   
   v=0
   o=- 3341834898 3341834898 IN IP4 Carrier_IP
   s=-
   c=IN IP4 Carrier_IP
   t=0 0
   m=audio 29864 RTP/AVP 18 0 101
   a=rtpmap:101 telephone-event/8000
   a=ptime:20
   a=silenceSupp:off - - - -


On Sep 14, 2012, at 4:42 PM, Michael Collins <msc@freeswitch.org> wrote:

Mike, try comma separating the values instead of using two sets of {}:
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");

-MC

On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
I have that set in the bridge command however does not seem to be working as expected.

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");



On Sep 14, 2012, at 3:16 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

set {sip_require_timer=false} in your outbound calls or globally 


On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
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"

The ULC are stating we need to only have it in our supported and do not pass them a require.

Re-Invite from B-Leg to FS
   ------------------------------------------------------------------------
   INVITE sip:16025551212;phone-context=+1@FS_SERVER:5070 SIP/2.0
   Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
   Via: SIP/2.0/UDP B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
   Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
   Contact: <sip:18475551212@B-LEG_IP:5060;transport=udp>
   Content-Length: 217
   Content-Type: application/sdp
   CSeq: 33480808 INVITE
   From: <sip:18475551212@B-LEG_IP>;tag=100052073
   Max-Forwards: 92
   Session-Expires: 3600;refresher=uas
   Supported: timer
   To: <sip:116025551212@OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
   User-Agent: FreeSwitch

   v=0
   o=- 3308986892 0 IN IP4 B-LEG_IP
   s=Media Server
   c=IN IP4 B-LEG_IP
   t=0 0
   m=audio 51246 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   ------------------------------------------------------------------------

Re-Invite from FS to A-Leg
  ------------------------------------------------------------------------
   INVITE sip:16025551212@DID_CARRIER:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
   Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
   Max-Forwards: 97
   From: <sip:18475551212;phone-context=+1@OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
   To: <sip:16025551212;phone-context=+1@DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
   Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
   CSeq: 33480811 INVITE
   Contact: <sip:18475551212;phone-context=+1@FS_SERVER:5070;transport=udp>
   User-Agent: FreeSwitch
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
   Require: timer
   Supported: timer, precondition, path, replaces
   Session-Expires: 64800;refresher=uas
   Min-SE: 64800
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 235
   X-FS-Support: update_display,send_info

   v=0
   o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
   s=Media Server
   c=IN IP4 B-LEG_IP
   t=0 0
   m=audio 51246 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   ------------------------------------------------------------------------


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_________________________________________________________________________
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_________________________________________________________________________
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--
Michael S Collins
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_________________________________________________________________________
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