[Freeswitch-users] Outbound only calls don't connect when bypass_media is true.

Anthony Minessale anthony.minessale at gmail.com
Mon Jan 31 01:34:09 MSK 2011 

Just do not use bypass media.
That is all you can do in that situation.


On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <marcin321 at hotmail.com> wrote:
> I just want to add that I enabled STUN on my cell so now the SDP message in
> the INVITE to voip.ms contains the public IP of my phone, but it still
> doesn't work.
>
> ________________________________
> From: marcin321 at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 28 Jan 2011 19:54:19 -0500
> Subject: [Freeswitch-users] Outbound only calls don't connect when
> bypass_media is true.
>
> Hello,
>
> I'm a new user of freeswitch, so please bear with me. I have the
> following setup:
> voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP ->
> my nokia cellphone on AT&T wireless. This setup is intended to conserve the
> battery usage.
> I've managed to make everything work well when I'm calling in over any phone
> to my cell phone, and freeswitch is enabled to work in bypass_media = true,
> even though by cell is behind NAT on at&t's network. Things break when I
> pick up my cell and try to call my home phone (or any phone for that
> matter). This is the relevant snippet from my dialplan:
> <extension name="outbound">
>   <condition field="destination_number"
> expression="^1?([2-9]\d{2}[2-9]\d{6})$">
>     <!--<action application="set" data="bypass_media=true"/>-->
>     <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
>   </condition>
> </extension>
>
> Like shown above, my call will go to my home phone. When I uncomment the
> bypass_media tag, my call will not connect. Here are the siptraces
> I replaced my real home phone number in the with "MYPHONE".
>
> recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>    ------------------------------------------------------------------------
>    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>    To: <sip:MYPHONE at 192.168.1.100>
>    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
>    Supported: 100rel,timer
>    CSeq: 5244503 INVITE
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
>    User-Agent: S60 RM-624 v 20.2.042 (en)
>    Expires: 120
>    Privacy: None
>    Session-Expires: 1800
>    Max-Forwards: 70
>    Content-Type: application/sdp
>    Accept-Language: en
>    Content-Length: 292
>
>    v=0
>    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>    s=-
>    c=IN IP4 10.153.174.6
>    t=0 0
>    m=audio 49152 RTP/AVP 18 97 98
>    a=sendrecv
>    a=rtpmap:18 G729/8000
>    a=ptime:20
>    a=maxptime:40
>    a=fmtp:18 annexb=no
>    a=rtpmap:97 iLBC/8000
>    a=rtpmap:98 telephone-event/8000
>    a=fmtp:98 0-15
>    ------------------------------------------------------------------------
> send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>    To: <sip:MYPHONE at 192.168.1.100>
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    CSeq: 5244503 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>    ------------------------------------------------------------------------
>    SIP/2.0 407 Proxy Authentication Required
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on
> sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip 32.136.78.180
>
>    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    CSeq: 5244503 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Proxy-Authenticate: Digest realm="192.168.1.100",
> nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
>    ------------------------------------------------------------------------
>    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    CSeq: 5244503 ACK
>    Supported: sec-agree
>    Max-Forwards: 70
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
>    ------------------------------------------------------------------------
>    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>    To: <sip:MYPHONE at 192.168.1.100>
>    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
>    Supported: 100rel,timer
>    CSeq: 5244504 INVITE
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
>    User-Agent: S60 RM-624 v 20.2.042 (en)
>    Expires: 120
>    Privacy: None
>    Session-Expires: 1800
>    Max-Forwards: 70
>    Proxy-Authorization: Digest
> qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
>    Content-Type: application/sdp
>    Accept-Language: en
>    Content-Length: 292
>
>    v=0
>    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>    s=-
>    c=IN IP4 10.153.174.6
>    t=0 0
>    m=audio 49152 RTP/AVP 18 97 98
>    a=sendrecv
>    a=rtpmap:18 G729/8000
>    a=ptime:20
>    a=maxptime:40
>    a=fmtp:18 annexb=no
>    a=rtpmap:97 iLBC/8000
>    a=rtpmap:98 telephone-event/8000
>    a=fmtp:98 0-15
>    ------------------------------------------------------------------------
> send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>    To: <sip:MYPHONE at 192.168.1.100>
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    CSeq: 5244504 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
> <1001>->MYPHONE in context default
> 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
>    ------------------------------------------------------------------------
>    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
>    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
>    Max-Forwards: 69
>    From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>    To: <sip:1MYPHONE at newyork.voip.ms>
>    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>    CSeq: 7788615 INVITE
>    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 280
>    X-FS-Support: update_display
>    Remote-Party-ID: "Extension 1001"
> <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>    s=-
>    c=IN IP4 10.153.174.6
>    t=0 0
>    m=audio 49152 RTP/AVP 18 97 98
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:97 iLBC/8000
>    a=rtpmap:98 telephone-event/8000
>    a=fmtp:98 0-15
>    a=ptime:20
>    a=maxptime:40
>    ------------------------------------------------------------------------
> recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
>    ------------------------------------------------------------------------
>    SIP/2.0 407 Proxy Authentication Required
>    Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
>    From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
>    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>    CSeq: 7788615 INVITE
>    User-Agent: VoIPMS/SERAST
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>    Supported: replaces
>    Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> nonce="2d534dd6"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
>    ------------------------------------------------------------------------
>    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
>    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
>    Max-Forwards: 69
>    From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
>    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>    CSeq: 7788615 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
>    ------------------------------------------------------------------------
>    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
>    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
>    Max-Forwards: 69
>    From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>    To: <sip:1MYPHONE at newyork.voip.ms>
>    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>    CSeq: 7788616 INVITE
>    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
>    Expires: 300
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Proxy-Authorization: Digest username="121628", realm="newyork.voip.ms",
> nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> response="16f3301efae13df926da7550f709d28a"
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 280
>    X-FS-Support: update_display
>    Remote-Party-ID: "Extension 1001"
> <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>    s=-
>    c=IN IP4 10.153.174.6
>    t=0 0
>    m=audio 49152 RTP/AVP 18 97 98
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:97 iLBC/8000
>    a=rtpmap:98 telephone-event/8000
>    a=fmtp:98 0-15
>    a=ptime:20
>    a=maxptime:40
>    ------------------------------------------------------------------------
> recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
>    From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>    To: <sip:1MYPHONE at newyork.voip.ms>
>    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>    CSeq: 7788616 INVITE
>    User-Agent: VoIPMS/SERAST
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>    Supported: replaces
>    Contact: <sip:1MYPHONE at 74.63.41.218>
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
>    ------------------------------------------------------------------------
>    SIP/2.0 503 Service Unavailable
>    Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
>    From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
>    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>    CSeq: 7788616 INVITE
>    User-Agent: VoIPMS/SERAST
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>    Supported: replaces
>    Contact: <sip:1MYPHONE at 74.63.41.218>
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
>    ------------------------------------------------------------------------
>    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
>    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
>    Max-Forwards: 69
>    From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
>    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>    CSeq: 7788616 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
> Cause: NO_ANSWER
> 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> instruction, hanging up.
> 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
> (sofia/external/1MYPHONE) Ended
> 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close Channel
> sofia/external/1MYPHONE [CS_DESTROY]
> send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
>    ------------------------------------------------------------------------
>    SIP/2.0 503 Service Unavailable
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    CSeq: 5244504 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
> (sofia/internal/1001 at 192.168.1.100) Ended
>    Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> switch_core_session.c:1308 Close Channel sofia/internal/1001 at 192.168.1.100
> [CS_DESTROY]
>
>    Remote-Party-ID: "MYPHONE"
> <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
>
>    ------------------------------------------------------------------------
> recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
>    ------------------------------------------------------------------------
>    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>    Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
>    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
>    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>    CSeq: 5244504 ACK
>    Supported: sec-agree
>    Max-Forwards: 70
>    Proxy-Authorization: Digest
> qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
> Thank you in advance.
>
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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 at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

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