Require Timer on re-invite from FS to A-LEG
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 ------------------------------------------------------------------------
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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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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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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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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If its b leg sending invite you need to set it on that leg, so in this case set it on the so too or just globally set it in vars.xml and forget about it and it will apply to all legs which should be ok. This particular case is a bot of a philosophical debate on proper action as we have devices break on it going both ways hence the setting. On Sep 14, 2012 6:44 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
------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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gonna set it in vars.xml and call it a day and see if it works On Sep 14, 2012, at 5:16 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
If its b leg sending invite you need to set it on that leg, so in this case set it on the so too or just globally set it in vars.xml and forget about it and it will apply to all legs which should be ok. This particular case is a bot of a philosophical debate on proper action as we have devices break on it going both ways hence the setting.
On Sep 14, 2012 6:44 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 ------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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 ------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
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FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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: ------------------------------------------------------------------------ INVITE sip:18665551212@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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
------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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 Both carriers use Acme Packet as the SBC and I have a feeling this is where the hangup is coming in at 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: ------------------------------------------------------------------------ INVITE sip:18665551212@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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 ------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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However and oddly enough Level(3)'s Sonus platform deals with this just fine On Sep 14, 2012, at 5:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
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
Both carriers use Acme Packet as the SBC and I have a feeling this is where the hangup is coming in at
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: ------------------------------------------------------------------------ INVITE sip:18665551212@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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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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@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: ------------------------------------------------------------------------ INVITE sip:18665551212@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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 ------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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: ------------------------------------------------------------------------ INVITE sip:18665551212@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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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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<http://tools.ietf.org/html/rfc4028#ref-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:
------------------------------------------------------------------------ INVITE sip:18665551212@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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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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: ------------------------------------------------------------------------ INVITE sip:18665551212@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@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 Contact: <sip:19999218665551212@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@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 Contact: <sip:18665551212@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@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 >> ------------------------------------------------------------------------ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
-
Anthony Minessale
-
Michael Collins
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Mike Burlingame