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

Mike Burlingame mike.burlingame at me.com
Sat Sep 15 04:47:41 MSD 2012 

This worked when set in the vars.xml if you can please let me know on monday what you find that would be great for the time being I am going to keep pushing GC and PaeTec to fix it's config and or work with it's vendors.

On Sep 14, 2012, at 5:29 PM, Anthony Minessale <anthony.minessale at 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 at 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:
>    ------------------------------------------------------------------------
>    INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>    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 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>    To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>    CSeq: 219015274 INVITE
>    Expires: 180
>    Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>    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 at 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 at ULC>;tag=aXFgDty84N6gK
>    To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>    Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>    CSeq: 33497250 INVITE
>    Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>    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 at 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."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>> 
>> -MC
>> 
>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> I have that set in the bridge command however does not seem to be working as expected.
>> 
>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$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 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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>>> 
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>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
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>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
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>> 
>> -- 
>> Michael S Collins
>> Twitter: @mercutioviz
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>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
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>> http://www.freeswitchsolutions.com
>> 
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> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> http://wiki.freeswitch.org
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> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
> 
> 
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> 
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