[Freeswitch-users] Require Timer on re-invite from FS to A-LEG

[Mike Burlingame]

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 at 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 at 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 at 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 at B-LEG_IP:5060;transport=udp>
>    Content-Length: 217
>    Content-Type: application/sdp
>    CSeq: 33480808 INVITE
>    From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>    Max-Forwards: 92
>    Session-Expires: 3600;refresher=uas
>    Supported: timer
>    To: <sip:116025551212 at 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 at 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 at OpenSIPS_DID_CARRIER_Proxy:5060;user=phone>;tag=c9Favaa53XFXB
>    To: <sip:16025551212;phone-context=+1 at DID_CARRIER:5060;user=phone>;tag=SDd626401-gK095bbb72
>    Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>    CSeq: 33480811 INVITE
>    Contact: <sip:18475551212;phone-context=+1 at 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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> -- 
> Anthony Minessale II
> 
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