I agree they should not be dropping the call - was just passing on the info to complete out the thread as well as show it seems to be losing battle with the carriers to try to get them to change

Sent from my iPhone 4S

On Sep 17, 2012, at 8:11 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

Just because it says its not recommended does not mean it says you should intentionally drop a call because of it.
If you can make it work by turning off in the var then that is your solution.  As I have already said, there are several real-world situations where it breaks things going either way hence our solution was to make it work both ways.


On Mon, Sep 17, 2012 at 10:00 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
PaeTec and Global Crossing both feel that we should not be including the require based on the RFC posted below - I have read over the RFC and tend to agree based on my limited knowledge of the timer RFC

Based on what they are saying because the invite from them state that timers are supported that we should remember this and not be putting in a require

------------------
From carriers 
-------------------

Please see below…section 8.1 of RFC 4028

 

From RFC4028:

 

   The proxy MUST remember, for the duration of the transaction, whether

   the request contained the Supported header field with the value

   'timer'.  If the request did not contain a Supported header field

   with the value 'timer', the proxy *MAY* insert a Require header field

   with the value 'timer' into the request.  However, this is*NOT

   RECOMMENDED*.  This allows the proxy to insist on a session timer for

   the session.  This header field is not needed if a Supported header

   field was in the request; in this case, the proxy would already be

   sure the session timer can be used for the session.

 

 

Section 7.1 of RFC 4028

 

A UAC that supports the session timer extension defined here MUST

   include a Supported header field in each request (except ACK),

   listing the option tag 'timer' [2].  It MUST do so even if the UAC is

   not requesting usage of the session timer for this session.

 

   The UAC MAY include a Require header field in the request with the

   value 'timer' to indicate that the UAS must support the session timer

   to participate in the session.  This does not mean that the UAC is

   requiring the UAS to perform the refreshes, only that it is requiring

   the UAS to support the extension.  In addition, the UAC MAY include a

   Proxy-Require header field in the request with the value 'timer' to

   indicate that proxies must support the session timer in order to

   correctly process the request.  However, usage of either Require or

   Proxy-Require by the UAC is NOT RECOMMENDED.  They are not needed,

   since the extension works even when only the UAC supports the

   extension.  The Supported header field containing 'timer' MUST still

   be included, even if the Require or Proxy-Require header fields are

   present containing 'timer'

 


Sent from my iPhone 4S

On Sep 14, 2012, at 5:29 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

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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