[Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true

2009-12-01 Thread Juan Backson
Hi,

I found that with bypass_media=true, freeswitch would change c= to FS's own
IP.

I think this is a misconfiguration.  Does anyone know what config could have
caused that?

thanks,
jb
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Re: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true

2009-12-01 Thread Juan Backson
In the following trace,102 is FS IP, 104 is calling party and 13 is
called party.

with bypass_media, FS still changesc=IN IP4 192.168.1.102

Any idea why?


freeswi...@localhost.localdomain> recv 951 bytes from
udp/[192.168.1.104]:5060 at 22:56:33.782715:
   
   INVITE sip:90964...@192.168.1.102  SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.104:5060
;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport
   From: 
>;tag=786224322
   To: >
   Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104
   CSeq: 37 INVITE
   Contact: 
   Content-Type: application/sdp
   Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
   Max-Forwards: 70
   Supported: 100rel, replaces
   User-Agent: SIPPER for PhonerLite
   Content-Length:   397

   v=0
   o=- 3393406017 0 IN IP4 192.168.1.104
   s=SIPPER for PhonerLite
   c=IN IP4 192.168.1.104
   t=0 0
   m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:2 G726-32/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:97 iLBC/8000
   a=rtpmap:110 speex/8000
   a=rtpmap:111 speex/16000
   a=rtpmap:9 G722/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=sendrecv
   
send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145:
   
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.104:5060
;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060
   From: 
>;tag=786224322
   To: >
   Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104
   CSeq: 37 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095
   Content-Length: 0

   
2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/phonerl...@192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273]
2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/
phonerl...@192.168.1.102 entering state [received][100]
2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP:
v=0
o=- 3393406017 0 IN IP4 192.168.1.104
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.104
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


EXECUTE sofia/internal/phonerl...@192.168.1.102 info()
2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/phonerl...@192.168.1.102]
Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Caller-Username: [PhonerLite]
Caller-Dialplan: [class4]
Caller-Caller-ID-Name: [PhonerLite]
Caller-Caller-ID-Number: [PhonerLite]
Caller-Network-Addr: [192.168.1.104]
Caller-Destination-Number: [90964111]
Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/phonerl...@192.168.1.102]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1259708193783162]
Caller-Channel-Created-Time: [1259708193783162]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [192.168.1.104]
variable_sip_received_port: [5060]
variable_sip_via_protocol: [udp]
variable_sip_from_user: [PhonerLite]
variable_sip_from_uri: [phonerl...@192.168.1.102]
variable_sip_from_host: [192.168.1.102]
variable_sip_from_user_stripped: [PhonerLite]
variable_sip_from_tag: [786224322]
variable_sofia_profile_name: [internal]
variable_sip_req_user: [90964111]
variable_sip_req_uri: [90964...@192.168.1.102]
variable_sip_req_host: [192.168.1.102]
variable_sip_to_user: [90964111]
variable_sip_to_uri: [90964...@192.168.1.102]
variable_sip_to_host: [192.168.1.102]
variable_sip_contact_port: [5060]
variable_sip_contact_uri: [phonerl...@192.168.1.104:5060]
variable_sip_contact_host: [192.168.1.104]
variable_channel_name: [sofia/internal/phonerl...@192.168.1.102]
variable_sip_call_id: [003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104]
variable_sip_user_agent: [SIPPER for PhonerLite]
variable_sip_via_host: [192.168.1.104]
variable_sip_via_port: [5060]
variable_bypass_media: [true]
variable_proxy_media: [true]
variable_sip_via_rport: [5060]
variable_max_forwards: [70]
variable_switch_r_sdp: [v=0
o=- 3393406017 0 IN IP4 192.168.1.104
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.104
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000

Re: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true

2009-12-01 Thread Michael Jerris
The only way this would happen would be if this is set to proxy media  
not bypass.  Are you setting both?


Mike

On Dec 1, 2009, at 10:08 AM, Juan Backson  wrote:

In the following trace,102 is FS IP, 104 is calling party and 13  
is called party.


with bypass_media, FS still changesc=IN IP4 192.168.1.102

Any idea why?


freeswi...@localhost.localdomain> recv 951 bytes from udp/ 
[192.168.1.104]:5060 at 22:56:33.782715:

--- 
-

   INVITE sip:90964...@192.168.1.102 SIP/2.0
   Via: SIP/2.0/UDP  
192.168.1.104: 
5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport

   From: ;tag=786224322
   To: 
   Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104
   CSeq: 37 INVITE
   Contact: 
   Content-Type: application/sdp
   Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE,  
UPDATE

   Max-Forwards: 70
   Supported: 100rel, replaces
   User-Agent: SIPPER for PhonerLite
   Content-Length:   397

   v=0
   o=- 3393406017 0 IN IP4 192.168.1.104
   s=SIPPER for PhonerLite
   c=IN IP4 192.168.1.104
   t=0 0
   m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:2 G726-32/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:97 iLBC/8000
   a=rtpmap:110 speex/8000
   a=rtpmap:111 speex/16000
   a=rtpmap:9 G722/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=sendrecv

--- 
-

send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145:

--- 
-

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP  
192.168.1.104: 
5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060

   From: ;tag=786224322
   To: 
   Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104
   CSeq: 37 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095
   Content-Length: 0


--- 
-
2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel  
sofia/internal/phonerl...@192.168.1.102 [d4233c9a- 
ee3b-40d4-910d-3b1579f9a273]
2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/ 
internal/phonerl...@192.168.1.102 entering state [received][100]

2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP:
v=0
o=- 3393406017 0 IN IP4 192.168.1.104
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.104
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


EXECUTE sofia/internal/phonerl...@192.168.1.102 info()
2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/phonerl...@192.168.1.102]
Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Caller-Username: [PhonerLite]
Caller-Dialplan: [class4]
Caller-Caller-ID-Name: [PhonerLite]
Caller-Caller-ID-Number: [PhonerLite]
Caller-Network-Addr: [192.168.1.104]
Caller-Destination-Number: [90964111]
Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/phonerl...@192.168.1.102]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1259708193783162]
Caller-Channel-Created-Time: [1259708193783162]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [192.168.1.104]
variable_sip_received_port: [5060]
variable_sip_via_protocol: [udp]
variable_sip_from_user: [PhonerLite]
variable_sip_from_uri: [phonerl...@192.168.1.102]
variable_sip_from_host: [192.168.1.102]
variable_sip_from_user_stripped: [PhonerLite]
variable_sip_from_tag: [786224322]
variable_sofia_profile_name: [internal]
variable_sip_req_user: [90964111]
variable_sip_req_uri: [90964...@192.168.1.102]
variable_sip_req_host: [192.168.1.102]
variable_sip_to_user: [90964111]
variable_sip_to_uri: [90964...@192.168.1.102]
variable_sip_to_host: [192.168.1.102]
variable_sip_contact_port: [5060]
variable_sip_contact_uri: [phonerl...@192.168.1.104:5060]
variable_sip_contact_host: [192.168.1.104]
variable_channel_name: [sofia/internal/phonerl...@192.168.1.102]
variable_sip_call_id: [003C8E1B-F8DC-DE11- 
a853-001a80565...@192.168.1.104]

variable_sip_user_agent: [SIPPER for PhonerLite]
variable_sip_via_host: [192.168.1.104]
variable_sip_via_port: [5060]
variable_bypass_media: [true]
variable_proxy_media: [true]
variable_sip_via_rport: [5060]
variable_m

Re: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true

2009-12-01 Thread Anthony Minessale
yes he did you can see it in his trace.
you can not use both of them together..


On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris  wrote:

> The only way this would happen would be if this is set to proxy media not
> bypass.  Are you setting both?
>
> Mike
>
> On Dec 1, 2009, at 10:08 AM, Juan Backson  wrote:
>
> In the following trace,102 is FS IP, 104 is calling party and 13 is
> called party.
>
> with bypass_media, FS still changesc=IN IP4 192.168.1.102
>
> Any idea why?
>
>
> freeswi...@localhost.localdomain> recv 951 bytes from
> udp/[192.168.1.104]:5060 at 22:56:33.782715:
>
>INVITE sip:90964...@192.168.1.102 SIP/2.0
>Via: SIP/2.0/UDP 192.168.1.104:5060
> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport
>From: 
> >;tag=786224322
>To: >
>Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104
>CSeq: 37 INVITE
>Contact: 
>Content-Type: application/sdp
>Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
>Max-Forwards: 70
>Supported: 100rel, replaces
>User-Agent: SIPPER for PhonerLite
>Content-Length:   397
>
>v=0
>o=- 3393406017 0 IN IP4 192.168.1.104
>s=SIPPER for PhonerLite
>c=IN IP4 192.168.1.104
>t=0 0
>m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
>a=rtpmap:8 PCMA/8000
>a=rtpmap:0 PCMU/8000
>a=rtpmap:2 G726-32/8000
>a=rtpmap:3 GSM/8000
>a=rtpmap:97 iLBC/8000
>a=rtpmap:110 speex/8000
>a=rtpmap:111 speex/16000
>a=rtpmap:9 G722/8000
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-16
>a=sendrecv
>
> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145:
>
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 192.168.1.104:5060
> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060
>From: 
> >;tag=786224322
>To: >
>Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104
>CSeq: 37 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095
>Content-Length: 0
>
>
> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/phonerl...@192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273]
> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/
> phonerl...@192.168.1.102 entering state [received][100]
> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP:
> v=0
> o=- 3393406017 0 IN IP4 192.168.1.104
> s=SIPPER for PhonerLite
> c=IN IP4 192.168.1.104
> t=0 0
> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:111 speex/16000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
>
> EXECUTE sofia/internal/phonerl...@192.168.1.102 info()
> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [sofia/internal/phonerl...@192.168.1.102]
> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Answer-State: [ringing]
> Caller-Username: [PhonerLite]
> Caller-Dialplan: [class4]
> Caller-Caller-ID-Name: [PhonerLite]
> Caller-Caller-ID-Number: [PhonerLite]
> Caller-Network-Addr: [192.168.1.104]
> Caller-Destination-Number: [90964111]
> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273]
> Caller-Source: [mod_sofia]
> Caller-Context: [default]
> Caller-Channel-Name: [sofia/internal/phonerl...@192.168.1.102]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1259708193783162]
> Caller-Channel-Created-Time: [1259708193783162]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_sip_received_ip: [192.168.1.104]
> variable_sip_received_port: [5060]
> variable_sip_via_protocol: [udp]
> variable_sip_from_user: [PhonerLite]
> variable_sip_from_uri: [phonerl...@192.168.1.102]
> variable_sip_from_host: [192.168.1.102]
> variable_sip_from_user_stripped: [PhonerLite]
> variable_sip_from_tag: [786224322]
> variable_sofia_profile_name: [internal]
> variable_sip_req_user: [90964111]
> variable_sip_req_uri: [90964...@192.168.1.102]
> variable_sip_req_host: [192.168.1.102]
> variable_sip_to_user: [90964111]
> variable_sip_to_uri: [90964...@192.168.1.102]
> variable_sip_to_host: [192.168.1.102]
> variable_sip_contact_port: [5060]
> variable_sip_contact_uri: [phonerl...@192.168.1.104:5060]
> variable_sip_contact_host: [192.168.1