Hi all, I have configure freeswitch as SBC server with one asterisk server. I am getting nat issue when make call from FS to FS. my case as: client --- FS ----- Asterisk server ----- FS -----client. my configure in public.xml like: <extension name="asteirsk"> <condition field="destination_number" expression="(\d+)$" require-nested="true"> <condition field="network_addr" expression="^192\.168\.1\.123$" break="never"> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/${destination_number}@${domain_name}"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <anti-action application="set" data="bypass_media=true"/> <anti-action application="bridge" data="sofia/internal/${destination_number}@192.168.1.123"/> </condition> </condition> </extension> Call is process ok, But I see that when I answer call in destination, asterisk server alway send 200OK to my FS, FS return ACk, but asterisk resend 200OK. if any one can help, thank all alot. I am getting this stage to long and cant find any solution here. Thanks, ThanhTruong
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason? On Sep 9, 2013, at 5:31 AM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
Hi all,
I have configure freeswitch as SBC server with one asterisk server.
I am getting nat issue when make call from FS to FS.
my case as:
client --- FS ----- Asterisk server ----- FS -----client.
my configure in public.xml like:
<extension name="asteirsk"> <condition field="destination_number" expression="(\d+)$" require-nested="true"> <condition field="network_addr" expression="^192\.168\.1\.123$" break="never"> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/${destination_number}@${domain_name}"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <anti-action application="set" data="bypass_media=true"/> <anti-action application="bridge" data="sofia/internal/${destination_number}@192.168.1.123"/> </condition> </condition> </extension>
Call is process ok, But I see that when I answer call in destination, asterisk server alway send 200OK to my FS, FS return ACk, but asterisk resend 200OK.
if any one can help, thank all alot. I am getting this stage to long and cant find any solution here.
Thanks,
ThanhTruong
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hi this is sip trace in FS server. ThanhTruong ________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason? On Sep 9, 2013, at 5:31 AM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
Hi all,
I have configure freeswitch as SBC server with one asterisk server.
I am getting nat issue when make call from FS to FS.
my case as:
client --- FS ----- Asterisk server ----- FS -----client.
my configure in public.xml like:
<extension name="asteirsk"> <condition field="destination_number" expression="(\d+)$" require-nested="true"> <condition field="network_addr" expression="^192\.168\.1\.123$" break="never"> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/${destination_number}@${domain_name}"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <anti-action application="set" data="bypass_media=true"/> <anti-action application="bridge" data="sofia/internal/${destination_number}@192.168.1.123"/> </condition> </condition> </extension>
Call is process ok, But I see that when I answer call in destination, asterisk server alway send 200OK to my FS, FS return ACk, but asterisk resend 200OK.
if any one can help, thank all alot. I am getting this stage to long and cant find any solution here.
Thanks,
ThanhTruong
_________________________________________________________________________ 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
Hi, this is sip trace in FS server. interface: eth0 (123.123.123.127/255.255.255.224) filter: (ip) and ( port 5060 ) U 123.21.238.196:22938 -> 123.123.123.123:5060 INVITE sip:1001@123.123.123.123 SIP/2.0. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d61b031c5e253c03-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1000@123.21.238.196:22938>. To: "1001"<sip:1001@123.123.123.123>. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 475. . v=0. o=- 2 2 IN IP4 123.21.238.196. s=CounterPath X-Lite 3.0. c=IN IP4 123.21.238.196. t=0 0. m=audio 45146 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 2 : +5fil0fi ho2/O/qw 192.168.1.2 45146. a=alt:2 1 : Ry0ImV0p jCU6QTba 169.254.186.19 45146. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d61b031c5e253c03-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001"<sip:1001@123.123.123.123>. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Content-Length: 0. . U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d61b031c5e253c03-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001"<sip:1001@123.123.123.123>. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 1 INVITE. Contact: <sip:1000@123.123.123.123>. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Length: 0. . U 123.123.123.123:5060 -> 123.123.123.126:5060 INVITE sip:1001@123.123.123.126 SIP/2.0. Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bK16gBQmKHS6Smj. Max-Forwards: 70. From: <sip:1000@123.123.123.126>;tag=DKcv9cF2pa3vr. To: <sip:1001@123.123.123.126>. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 2 INVITE. Contact: <sip:1000@123.123.123.126:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 157. . v=0. o=FreeSWITCH 1376087327 1376087328 IN IP4 123.123.123.123. s=FreeSWITCH. t=0 0. m=audio 8000 RTP/AVP 8 0. c=IN IP4 123.123.123.123. a=rtpmap:0 PCMU/8000. U 123.123.123.126:5060 -> 123.123.123.123:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK16gBQmKHS6Smj;received=123.123.123.123;rport=5060. From: <sip:1000@123.123.123.126>;tag=DKcv9cF2pa3vr. To: <sip:1001@123.123.123.126>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 2 INVITE. Server: Gentrice IPPBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="516141bc". Content-Length: 0. . U 123.123.123.123:5060 -> 123.123.123.126:5060 ACK sip:1001@123.123.123.126 SIP/2.0. Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bK16gBQmKHS6Smj. Max-Forwards: 70. From: <sip:1000@123.123.123.126>;tag=DKcv9cF2pa3vr. To: <sip:1001@123.123.123.126>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 2 ACK. Content-Length: 0. . U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d61b031c5e253c03-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001" <sip:1001@123.123.123.123>;tag=caK37HyyS1caD. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. WWW-Authenticate: Digest realm="asterisk", nonce="516141bc", algorithm=MD5. Content-Length: 0. . U 123.21.238.196:22938 -> 123.123.123.123:5060 ACK sip:1001@123.123.123.123 SIP/2.0. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d61b031c5e253c03-1--d87543-;rport. To: "1001" <sip:1001@123.123.123.123>;tag=caK37HyyS1caD. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 1 ACK. Content-Length: 0. . U 123.21.238.196:22938 -> 123.123.123.123:5060 INVITE sip:1001@123.123.123.123 SIP/2.0. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d04923161d7eba2b-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1000@123.21.238.196:22938>. To: "1001"<sip:1001@123.123.123.123>. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1011s stamp 41150. Authorization: Digest username="1000",realm="asterisk",nonce="516141bc",uri="sip:1001@123.123.123.123",response="f2a07bf5213207b0ffc682095868ce47",algorithm=MD5. Content-Length: 475. . v=0. o=- 2 2 IN IP4 123.21.238.196. s=CounterPath X-Lite 3.0. c=IN IP4 123.21.238.196. t=0 0. m=audio 45146 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 2 : +5fil0fi ho2/O/qw 192.168.1.2 45146. a=alt:2 1 : Ry0ImV0p jCU6QTba 169.254.186.19 45146. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d04923161d7eba2b-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001"<sip:1001@123.123.123.123>. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 2 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Content-Length: 0. . U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d04923161d7eba2b-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001" <sip:1001@123.123.123.123>;tag=ev5mB8Z5KKSFm. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 2 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Proxy-Authenticate: Digest realm="123.123.123.123", nonce="f8e49022-48fa-4f40-9598-9ba8c910768f", algorithm=MD5, qop="auth". Content-Length: 0. . U 123.21.238.196:22938 -> 123.123.123.123:5060 ACK sip:1001@123.123.123.123 SIP/2.0. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-d04923161d7eba2b-1--d87543-;rport. To: "1001" <sip:1001@123.123.123.123>;tag=ev5mB8Z5KKSFm. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 2 ACK. Content-Length: 0. . U 123.21.238.196:22938 -> 123.123.123.123:5060 INVITE sip:1001@123.123.123.123 SIP/2.0. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-6f33d61ceb017321-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1000@123.21.238.196:22938>. To: "1001"<sip:1001@123.123.123.123>. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1000",realm="123.123.123.123",nonce="f8e49022-48fa-4f40-9598-9ba8c910768f",uri="sip:1001@123.123.123.123",response="c462ce26b110d412f825fe0577c63453",cnonce="2f3a630b7a6dfdb967e55008c6f55265",nc=00000001,qop=auth,algorithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Authorization: Digest username="1000",realm="asterisk",nonce="516141bc",uri="sip:1001@123.123.123.123",response="f2a07bf5213207b0ffc682095868ce47",algorithm=MD5. Content-Length: 475. . v=0. o=- 2 2 IN IP4 123.21.238.196. s=CounterPath X-Lite 3.0. c=IN IP4 123.21.238.196. t=0 0. m=audio 45146 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 2 : +5fil0fi ho2/O/qw 192.168.1.2 45146. a=alt:2 1 : Ry0ImV0p jCU6QTba 169.254.186.19 45146. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtp U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-6f33d61ceb017321-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001"<sip:1001@123.123.123.123>. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Content-Length: 0. . U 123.123.123.123:5060 -> 123.123.123.126:5060 INVITE sip:1001@123.123.123.126 SIP/2.0. Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bK2Fa4rF4mpFg7D. Max-Forwards: 69. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126:5060>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Contact: <sip:1000@123.123.123.123:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Authorization: Digest username="1000", realm="asterisk", uri="sip:1001@123.123.123.123", nonce="516141bc", response="f2a07bf5213207b0ffc682095868ce47", algorithm=MD5. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 463. X-ROUTE: LOOKUP. X-FS-Support: update_display,send_info. Remote-Party-ID: "1000" <sip:1000@123.123.123.123>;party=calling;screen=yes;privacy=off. . v=0. o=- 2 2 IN IP4 123.21.238.196. s=CounterPath X-Lite 3.0. c=IN IP4 123.21.238.196. t=0 0. m=audio 45146 RTP/AVP 107 119 100 106 0 105 98 8 101. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=alt:1 2 : +5fil0fi ho2/O/qw 192.168.1.2 45146. a=alt:2 1 : Ry0ImV0p jCU6QTba 169.254.186.19 45146. U 123.123.123.126:5060 -> 123.123.123.123:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK2Fa4rF4mpFg7D;received=123.123.123.123;rport=5060. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126:5060>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Server: Gentrice IPPBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1001@123.123.123.126:5060>. Content-Length: 0. . U 123.123.123.126:5060 -> 123.123.123.123:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK2Fa4rF4mpFg7D;received=123.123.123.123;rport=5060. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126:5060>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Server: Gentrice IPPBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1001@123.123.123.126:5060>. Content-Length: 0. . U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-6f33d61ceb017321-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001" <sip:1001@123.123.123.123>;tag=F5yDD3g9gvF2F. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Contact: <sip:1001@123.123.123.123:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Length: 0. Remote-Party-ID: "Outbound Call" <sip:1001@123.123.123.123>;party=calling;privacy=off;screen=no. . U 123.123.123.126:5060 -> 123.123.123.123:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK2Fa4rF4mpFg7D;received=123.123.123.123;rport=5060. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126:5060>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Server: Gentrice IPPBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1001@123.123.123.126:5060>. Content-Type: application/sdp. Content-Length: 262. . v=0. o=gentrice 1875783318 1875783318 IN IP4 123.123.123.126. s=Gentrice IPPBX. c=IN IP4 123.123.123.126. t=0 0. m=audio 18668 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 123.123.123.123:5060 -> 123.123.123.126:5060 ACK sip:1001@123.123.123.126:5060 SIP/2.0. Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bK3r3vtaNrKr6SS. Max-Forwards: 70. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 ACK. Contact: <sip:1000@123.123.123.123:5060>. Authorization: Digest username="1000", realm="asterisk", uri="sip:1001@123.123.123.123", nonce="516141bc", response="f2a07bf5213207b0ffc682095868ce47", algorithm=MD5. Content-Length: 0. . U 123.123.123.123:5060 -> 123.21.238.196:22938 SIP/2.0 200 OK. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-6f33d61ceb017321-1--d87543-;rport=22938. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. To: "1001" <sip:1001@123.123.123.123>;tag=F5yDD3g9gvF2F. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Contact: <sip:1001@123.123.123.123:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.4.0. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 250. Remote-Party-ID: "Outbound Call" <sip:1001@123.123.123.123>;party=calling;privacy=off;screen=no. . v=0. o=gentrice 1875783318 1875783318 IN IP4 123.123.123.126. s=Gentrice IPPBX. c=IN IP4 123.123.123.126. t=0 0. m=audio 18668 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 123.21.238.196:22938 -> 123.123.123.123:5060 ACK sip:1001@123.123.123.123:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 123.21.238.196:22938;branch=z9hG4bK-d87543-b84af33f8130dd5e-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1000@123.21.238.196:22938>. To: "1001"<sip:1001@123.123.123.123>;tag=F5yDD3g9gvF2F. From: "1000"<sip:1000@123.123.123.123>;tag=eb72d35a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 ACK. Proxy-Authorization: Digest username="1000",realm="123.123.123.123",nonce="f8e49022-48fa-4f40-9598-9ba8c910768f",uri="sip:1001@123.123.123.123",response="c462ce26b110d412f825fe0577c63453",cnonce="2f3a630b7a6dfdb967e55008c6f55265",nc=00000001,qop=auth,algorithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Authorization: Digest username="1000",realm="asterisk",nonce="516141bc",uri="sip:1001@123.123.123.123",response="f2a07bf5213207b0ffc682095868ce47",algorithm=MD5. Content-Length: 0. . U 123.123.123.126:5060 -> 123.123.123.123:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK2Fa4rF4mpFg7D;received=123.123.123.123;rport=5060. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126:5060>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Server: Gentrice IPPBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1001@123.123.123.126:5060>. Content-Type: application/sdp. Content-Length: 262. . v=0. o=gentrice 1875783318 1875783318 IN IP4 123.123.123.126. s=Gentrice IPPBX. c=IN IP4 123.123.123.126. t=0 0. m=audio 18668 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 123.123.123.123:5060 -> 123.123.123.126:5060 ACK sip:1001@123.123.123.126:5060 SIP/2.0. Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bK3r3vtaNrKr6SS. Max-Forwards: 70. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 ACK. Contact: <sip:1000@123.123.123.123:5060>. Authorization: Digest username="1000", realm="asterisk", uri="sip:1001@123.123.123.123", nonce="516141bc", response="f2a07bf5213207b0ffc682095868ce47", algorithm=MD5. Content-Length: 0. . U 123.123.123.126:5060 -> 123.123.123.123:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK2Fa4rF4mpFg7D;received=123.123.123.123;rport=5060. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126:5060>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Server: Gentrice IPPBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1001@123.123.123.126:5060>. Content-Type: application/sdp. Content-Length: 262. . v=0. o=gentrice 1875783318 1875783318 IN IP4 123.123.123.126. s=Gentrice IPPBX. c=IN IP4 123.123.123.126. t=0 0. m=audio 18668 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 123.123.123.123:5060 -> 123.123.123.126:5060 ACK sip:1001@123.123.123.126:5060 SIP/2.0. Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bK3r3vtaNrKr6SS. Max-Forwards: 70. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 ACK. Contact: <sip:1000@123.123.123.123:5060>. Authorization: Digest username="1000", realm="asterisk", uri="sip:1001@123.123.123.123", nonce="516141bc", response="f2a07bf5213207b0ffc682095868ce47", algorithm=MD5. Content-Length: 0. . U 123.21.238.196:22938 -> 123.123.123.123:5060 . . .............. U 123.123.123.126:5060 -> 123.123.123.123:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK2Fa4rF4mpFg7D;received=123.123.123.123;rport=5060. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126:5060>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 INVITE. Server: Gentrice IPPBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1001@123.123.123.126:5060>. Content-Type: application/sdp. Content-Length: 262. . v=0. o=gentrice 1875783318 1875783318 IN IP4 123.123.123.126. s=Gentrice IPPBX. c=IN IP4 123.123.123.126. t=0 0. m=audio 18668 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 123.123.123.123:5060 -> 123.123.123.126:5060 ACK sip:1001@123.123.123.126:5060 SIP/2.0. Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bK3r3vtaNrKr6SS. Max-Forwards: 70. From: "1000" <sip:1000@123.123.123.123>;tag=ger6ey1ce55mB. To: <sip:1001@123.123.123.126>;tag=as2cd7f37a. Call-ID: Y2M1MWQ1MWFhMjJmYWEzMDgwYzY4N2U3ZjhhZmYxYzk.. CSeq: 3 ACK. Contact: <sip:1000@123.123.123.123:5060>. Authorization: Digest username="1000", realm="asterisk", uri="sip:1001@123.123.123.123", nonce="516141bc", response="f2a07bf5213207b0ffc682095868ce47", algorithm=MD5. Content-Length: 0. . It Always getting 200OK and ACK from my FS server to asterisk server. Hope some one can help me on this case. thanks, ThanhTruong ________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason? On Sep 9, 2013, at 5:31 AM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
Hi all,
I have configure freeswitch as SBC server with one asterisk server.
I am getting nat issue when make call from FS to FS.
my case as:
client --- FS ----- Asterisk server ----- FS -----client.
my configure in public.xml like:
<extension name="asteirsk"> <condition field="destination_number" expression="(\d+)$" require-nested="true"> <condition field="network_addr" expression="^192\.168\.1\.123$" break="never"> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/${destination_number}@${domain_name}"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <anti-action application="set" data="bypass_media=true"/> <anti-action application="bridge" data="sofia/internal/${destination_number}@192.168.1.123"/> </condition> </condition> </extension>
Call is process ok, But I see that when I answer call in destination, asterisk server alway send 200OK to my FS, FS return ACk, but asterisk resend 200OK.
if any one can help, thank all alot. I am getting this stage to long and cant find any solution here.
Thanks,
ThanhTruong
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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So did you do what I suggested then or not? On Sep 9, 2013, at 1:13 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi this is sip trace in FS server.
From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
hi Michael Jerris, I have sent in another email. Do you see it ? Thanks ThanhTruong ________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:21 AM Subject: Re: [Freeswitch-users] NAT issue with SBC So did you do what I suggested then or not? On Sep 9, 2013, at 1:13 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote: hi
this is sip trace in FS server.
________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
_________________________________________________________________________ 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
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
On Sep 9, 2013, at 1:45 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi Michael Jerris,
I have sent in another email. Do you see it ?
Thanks
ThanhTruong
From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:21 AM Subject: Re: [Freeswitch-users] NAT issue with SBC
So did you do what I suggested then or not?
On Sep 9, 2013, at 1:13 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi this is sip trace in FS server.
From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
_________________________________________________________________________ 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
Hi Michael Jerris, In Both site, ACK coming into asterisk server, then, after some second, asterisk try to send 200OK to FS and FS return ACK too. That is my case. sorry about that if i disturb you. Thanks, ThanhTruong ________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:49 AM Subject: Re: [Freeswitch-users] NAT issue with SBC have you lookedat a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason? On Sep 9, 2013, at 1:45 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote: hi Michael Jerris,
I have sent in another email. Do you see it ?
Thanks
ThanhTruong
________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:21 AM Subject: Re: [Freeswitch-users] NAT issue with SBC
So did you do what I suggested then or not?
On Sep 9, 2013, at 1:13 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi
this is sip trace in FS server.
________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
_________________________________________________________________________ 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
Then the asterisk box is not accepting the ack for some reason. On Sep 9, 2013, at 1:57 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
Hi Michael Jerris,
In Both site, ACK coming into asterisk server, then, after some second, asterisk try to send 200OK to FS and FS return ACK too.
That is my case.
sorry about that if i disturb you.
Thanks,
ThanhTruong
From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:49 AM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
On Sep 9, 2013, at 1:45 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi Michael Jerris,
I have sent in another email. Do you see it ?
Thanks
ThanhTruong
From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:21 AM Subject: Re: [Freeswitch-users] NAT issue with SBC
So did you do what I suggested then or not?
On Sep 9, 2013, at 1:13 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi this is sip trace in FS server.
From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
_________________________________________________________________________ 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
Hi Michael Jerris, I have tried many ways to see how asterisk reject ACK but cant see any issue in asterisk server. this is log in asterisk server when it getting 200OK. Hope you can help me more in this case. <--- Reliably Transmitting (NAT) to 123.123.123.123:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bKQ6S88v04UNUgp;received=123.123.123.123;rport=5060 From: "1000" <sip:1000@123.123.123.123>;tag=yDvgQBjFN9teg To: <sip:1001@59.120.208.196:5060>;tag=as7d0ee969 Call-ID: NDI5MzhjYjc2N2Y2NjM2OWZlYjM3YzlhODczZWRiMzQ. CSeq: 3 INVITE Server: Asterisk PBX 11.5.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:1001@123.123.123.124:5060> Content-Type: application/sdp Require: timer Content-Length: 304 v=0 o=root 1869340460 1869340460 IN IP4 123.123.123.124 s=Asterisk PBX 11.5.0 c=IN IP4 123.123.123.124 t=0 0 m=audio 14742 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 0 RTP/AVP 34 <------------> > 0x1fdb630 -- Probation passed - setting RTP source address to 123.123.123.125:34156 > 0x1fdb630 -- Probation passed - setting RTP source address to 123.123.123.125:34156 <--- SIP read from UDP:123.123.123.123:5060 ---> ACK sip:1001@123.123.123.124:5060 SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bKrFK1arH8ryH3H Max-Forwards: 70 From: "1000" <sip:1000@123.123.123.123>;tag=yDvgQBjFN9teg To: <sip:1001@123.123.123.124>;tag=as7d0ee969 Call-ID: NDI5MzhjYjc2N2Y2NjM2OWZlYjM3YzlhODczZWRiMzQ. CSeq: 3 ACK Contact: <sip:1000@123.123.123.123:5060> Authorization: Digest username="1000", realm="asterisk", uri="sip:1001@123.123.123.123", nonce="2d8e3c5a", response="afda149dd7fa2ba457ccc494c272a1a1", algorithm=MD5 Content-Length: 0 i make call from 123.123.123.125. FS server is 123.123.123.123. asterisk server is 123.123.123.124 Thank you in advance. ThanhTruong ________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 1:13 AM Subject: Re: [Freeswitch-users] NAT issue with SBC Then the asterisk box is not accepting the ack for some reason. On Sep 9, 2013, at 1:57 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote: Hi Michael Jerris,
In Both site, ACK coming into asterisk server, then, after some second, asterisk try to send 200OK to FS and FS return ACK too.
That is my case.
sorry about that if i disturb you.
Thanks, ThanhTruong
________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:49 AM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you lookedat a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
On Sep 9, 2013, at 1:45 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi Michael Jerris,
I have sent in another email. Do you see it ?
Thanks
ThanhTruong
________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, September 10, 2013 12:21 AM Subject: Re: [Freeswitch-users] NAT issue with SBC
So did you do what I suggested then or not?
On Sep 9, 2013, at 1:13 PM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
hi
this is sip trace in FS server.
________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 7:14 PM Subject: Re: [Freeswitch-users] NAT issue with SBC
have you looked at a Sip trace from both sides? Is the ACK getting to asterisk? Is it not accepting it for some reason?
_________________________________________________________________________ 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
_________________________________________________________________________ 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 (2)
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Michael Jerris
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ThanhTruong