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

30 Jan 2011

      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@hotmail.com
To: freeswitch-users@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@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@192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE@192.168.1.100>

   Contact: <sip:M9jdt73ig0oOJSbt6Uyy@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@192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE@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@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@192.168.1.100] from ip 
32.136.78.180

   To: <sip:MYPHONE@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@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@192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE@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@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@192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE@192.168.1.100>

   Contact: <sip:M9jdt73ig0oOJSbt6Uyy@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@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@192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE@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@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@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@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE@newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788615 INVITE

   Contact: <sip:gw+voip.ms@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@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@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE@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@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@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE@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@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@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE@newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 INVITE

   Contact: <sip:gw+voip.ms@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@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@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@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE@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@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@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE@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@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@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@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE@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@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@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@192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE@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@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@192.168.1.100 [CS_DESTROY]

   Remote-Party-ID: "MYPHONE" 
<sip:MYPHONE@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@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@192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE@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@192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"

   Content-Length: 0

   ------------------------------------------------------------------------

Thank you in advance.

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

Marcin Wojtowicz