Open Source Sip: Problems and solutions with ATLSIP

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96 Replies Last post: Jul 31, 2007 7:10 AM by Guest Go to original post 1 2 3 4 5 ... 7 Previous Next

tomach

64 posts since
Jun 4, 2007

30. Jun 20, 2007 4:32 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hello!

Ok call connections works fine etc...
I would like to ask how your activex works?
How your activex works? When call is created? is it working in separate thread then main winform application? Bacause I noticed that winforms do not hang and react normally so its ok (main thread is not stoped).

But my problem is that when I run your application (sample VB) under visual studio events come but they do NOT update controls. When I run the same application under cmd line everything works fine events come and update controls.

Do you had similar reaction under Visual Studio 2005?

I am really curious why is it happening? any suggestions?

Guest

31. Jun 20, 2007 6:41 AM

in response to: tomach

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hello.

tomach wrote:

Hmmm.

Yep. ATLSIP events come from a separate thread...

.NET is strict with regards to cross-thread UI updates (especially with
VS 2005). You may need to marshall UI updates to the UI thread for the
update to work.

For example with the ATLSIP_OnOutgoingCallTrying event, instead of
invoking lblStatus->Text directly from this separate thread you will
need to tell the UI thread to do it. Like this:

System::Void ATLSIP_OnOutgoingCallTrying(
System::Object^ sender,
AxATLSIPLib::_IOpenSIPStackCtlEvents_OnOutgoingCallTryingEvent^ e) {

// lblStatus->Text = L"Status: Trying"; // This does not work

SetText( lblStatus, L"Status: Trying" ); // This works.
}

delegate void SetTextCallback( Control^ control, String^ text);

void SetText( Control^ control, String^ text )
{
if ( control->InvokeRequired )
{
// We are NOT on the UI thread so we need to marshal
SetTextCallback^ settext = gcnew SetTextCallback( this,
&OSSPhone::Form1::SetText );
this->Invoke( settext, gcnew array{ control, text } );
} else
{
// We are on the UI thread. No need to marshal
control->Text = text;
}
}

The same logic applies with all other kinds of UI update. Anyway, this
is .NET-specific stuff. You can search for .NET InvokeRequired for
further reading.

Regards,
Ilian

I am really curious why is it happening? any suggestions?

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tomach

64 posts since
Jun 4, 2007

32. Jun 21, 2007 8:39 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hello!

Thanks a lot for the answer.
Right now I can register. The problem is that after I am registered and try to make call ATLSIP do not send "Proxy-Authorization" information:
for example:
Proxy-Authorization: Digest username="107", realm="192.168.2.69", nonce="00027556963e1f5da6b506f697a3907f2f53637b", uri="sip:00586820726@192.168.2.69", response="0e2bd6b0a8cc2c19a7a4ad744d4e017b", algorithm=MD5
Authentication Scheme: Digest
Username: "107"
Realm: "192.168.2.69"
Nonce Value: "00027556963e1f5da6b506f697a3907f2f53637b"
Authentication URI: "sip:00586820726@192.168.2.69"
Digest Authentication Response: "0e2bd6b0a8cc2c19a7a4ad744d4e017b"
Algorithm: MD5

Above is the example what is missing in INVITE that ALTSIP is sending to sipServer.
Is there any chance to add it, that INVITE would be complete?

Best Regards,
Tom

Guest

33. Jun 22, 2007 3:10 AM

in response to: tomach

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hi,

ATLSIP does reply with a Proxy-Authorization header in response to a
"407 Proxy Authentication Required" message.

Here are my logs from OSSPhone.

SEND: enc=0 747 Bytes to 192.168.0.107:11000:UDP (INVITE
sip:302@192.168.0.107 SIP/2.0) Interface Address=192.168.0.107
INVITE sip:302@192.168.0.107 SIP/2.0
From: 301 ;tag=5c022811fcf8181085d3df71ad1fd0c5
To: sip:302@192.168.0.107
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport
CSeq: 4711 INVITE
Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5
Contact: "301"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Type: application/sdp
Content-Length: 205

v=0
o=- 1182495843 1182495843 IN IP4 192.168.0.107
s=OSS RTP Session
c=IN IP4 192.168.0.107
t=0 0
m=audio 5000 RTP/AVP 101 4
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:4 G723/8000
----------------2:13:29.134----------------
RCV: enc=0 322 Bytes from RCVADDR: 192.168.0.107:RCVPORT: 11000:UDP
(SIP/2.0 100 Trying)
SIP/2.0 100 Trying
From: 301 ;tag=5c022811fcf8181085d3df71ad1fd0c5
To: sip:302@192.168.0.107
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107
CSeq: 4711 INVITE
Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5
Content-Length: 0

----------------2:13:29.216----------------
RCV: enc=0 567 Bytes from RCVADDR: 192.168.0.107:RCVPORT: 11000:UDP
(SIP/2.0 407 Proxy Authentication Required)
SIP/2.0 407 Proxy Authentication Required
From: 301 ;tag=5c022811fcf8181085d3df71ad1fd0c5
To: ;tag=b4512811fcf818109ff790ed26e13033
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport=5060;received=192.168.0.107
CSeq: 4711 INVITE
Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5
Proxy-Authenticate: Digest realm=192.168.0.107,
nonce="530e7db0dbb55accddef8327d684c2ca",
opaque="c9d379cc8f16ee0ac7fa8a1993816fc7", algorithm=MD5
Content-Length: 0

----------------2:13:29.257----------------
SEND: enc=0 543 Bytes to 192.168.0.107:11000:UDP (ACK
sip:302@192.168.0.107 SIP/2.0) Interface Address=192.168.0.107
ACK sip:302@192.168.0.107 SIP/2.0
From: 301 ;tag=5c022811fcf8181085d3df71ad1fd0c5
To: ;tag=b4512811fcf818109ff790ed26e13033
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport
CSeq: 4711 ACK
Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5
Contact: "301"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0

----------------2:13:29.240----------------
RCV: enc=0 567 Bytes from RCVADDR: 192.168.0.107:RCVPORT: 11000:UDP
(SIP/2.0 407 Proxy Authentication Required)
SIP/2.0 407 Proxy Authentication Required
From: 301 ;tag=5c022811fcf8181085d3df71ad1fd0c5
To: ;tag=b4512811fcf818109ff790ed26e13033
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport=5060;received=192.168.0.107
CSeq: 4711 INVITE
Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5
Proxy-Authenticate: Digest realm=192.168.0.107,
nonce="530e7db0dbb55accddef8327d684c2ca",
opaque="c9d379cc8f16ee0ac7fa8a1993816fc7", algorithm=MD5
Content-Length: 0

----------------2:13:29.310----------------
SEND: enc=0 945 Bytes to 192.168.0.107:11000:UDP (INVITE
sip:302@192.168.0.107 SIP/2.0) Interface Address=192.168.0.107
INVITE sip:302@192.168.0.107 SIP/2.0
From: 301 ;tag=5c022811fcf8181085d3df71ad1fd0c5
To: sip:302@192.168.0.107
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=2258;branch=z9hG4bK6e7c2811fcf8181085d3df71ad1fd0c5;uas-addr=192.168.0.107;rport
CSeq: 4712 INVITE
Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5
Contact: "301"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Proxy-Authorization: Digest username="301", realm="192.168.0.107",
nonce="530e7db0dbb55accddef8327d684c2ca", uri="sip:302@192.168.0.107",
response="2f4837e25f547ca1d87b548a0c232986", algorithm=MD5
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Type: application/sdp
Content-Length: 205

v=0
o=- 1182495843 1182495843 IN IP4 192.168.0.107
s=OSS RTP Session
c=IN IP4 192.168.0.107
t=0 0
m=audio 5000 RTP/AVP 101 4
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:4 G723/8000

tomach wrote:

Guest

34. Jun 22, 2007 3:51 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Ilian,

I am encountering clipping in the middle of the calls. what are the possible reasons? I will give you later the RTP

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Guest

35. Jun 22, 2007 4:55 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hi Andre,

This is probably caused by the adaptive silence detection algorithm.

Try using fixed silence detection. I''ve attached an email with the same
problem and a possible solution. You may have to tweak the threshold
value to find the sweet spot.

Regards,
Ilian

Andre Silo wrote:

Ilian,

I am encountering clipping in the middle of the calls. what are the possible reasons? I will give you later the RTP

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tomach

64 posts since
Jun 4, 2007

36. Jun 26, 2007 6:20 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hello!

Yes you are correct opensipsource resends proxy authorization but it is uncorrect.

I found an error in opensip.
Scenario:
Softphone''s ip is 192.168.2.45, user/password 507/507
SIpserver ip is 192.168.2.69

Problem:
I register wihtout any problem. But when I try to call:

In INVITE message which is send by the opensourcesip, field "From" looks like this:

From: 507 ;tag=6204057789f818108168ffe5741f66c1

where You can see should be address 192.168.2.69 because otherwise in proxy authentication required is uncorrect realm (should be 2.69 and tehre is 2.45).

When we look at the register request, we can see field from is correct:

From: ;tag=b94f4dd487f818108ff199e5a6d4f1ef

Only in invite it is uncorrect.

Have You met this probelm before?
If You need any additionl logs, I am ready to help.

Best Regards,
Tom

tomach

64 posts since
Jun 4, 2007

37. Jun 26, 2007 6:31 AM

in response to: tomach

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Instead of fields "From" above there should be:

In invite it looks like this:

From: 507 sip:507@192.168.2.45 ;tag=6204057789f818108168ffe5741f66c1

and for register:

From: sip:507@192.168.2.69 ;tag=b94f4dd487f818108ff199e5a6d4f1ef

Guest

38. Jun 26, 2007 6:58 AM

in response to: tomach

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hi Tom,

Hmmm. Can you check if you the latest versions of SoftphoneInterface.*
and Softphone.*?

This does not happen in my setup. My From headers are correct.

Here are my logs:

SEND: enc=0 531 Bytes to 70.42.73.140:5060:UDP (REGISTER
sip:70.42.73.140 SIP/2.0) Interface Address=
REGISTER sip:70.42.73.140 SIP/2.0
From: 300 ;tag=24b4836f04f9181094d7a2088e6bd734
To: sip:300@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1769;branch=z9hG4bK24b4836f04f9181094d8a2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 1 REGISTER
Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734
Contact: "300" <sip:300@192.168.0.107:5060;transport=udp>
User-Agent: OpenSIPStack-1.1.6-166
Expires: 3600
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0

----------------5:46:55.280----------------
RCV: enc=0 537 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP
(SIP/2.0 401 Unauthorized)
SIP/2.0 401 Unauthorized
From: 300 ;tag=24b4836f04f9181094d7a2088e6bd734
To: sip:300@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=3329;branch=z9hG4bK24b4836f04f9181094d8a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20
CSeq: 1 REGISTER
Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734
Server: OpenSIPStack-1.1.6-132
WWW-Authenticate: Digest realm="70.42.73.140",
nonce="0e73d21831c3e5cab945b3686a6f8fb3",
opaque="c307ecf61b2e21296250e3a528cdf064", algorithm=MD5
Content-Length: 0

----------------5:46:55.341----------------
SEND: enc=0 717 Bytes to 70.42.73.140:5060:UDP (REGISTER
sip:70.42.73.140 SIP/2.0) Interface Address=
REGISTER sip:70.42.73.140 SIP/2.0
From: 300 ;tag=48f7836f04f9181094d8a2088e6bd734
To: sip:300@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1769;branch=z9hG4bK48f7836f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 2 REGISTER
Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734
Contact: "300" <sip:300@192.168.0.107:5060;transport=udp>
User-Agent: OpenSIPStack-1.1.6-166
Expires: 3600
Max-Forwards: 10
Authorization: Digest username="300", realm="70.42.73.140",
nonce="0e73d21831c3e5cab945b3686a6f8fb3", uri="sip:70.42.73.140",
response="b85374c0f0b2adcdc6c983c70ff61dd4", algorithm=MD5
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0

----------------5:46:55.427----------------
RCV: enc=0 454 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP
(SIP/2.0 200 OK)
SIP/2.0 200 OK
From: 300 ;tag=48f7836f04f9181094d8a2088e6bd734
To: sip:300@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=3329;branch=z9hG4bK48f7836f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20
CSeq: 2 REGISTER
Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734
Contact: "300" <sip:300@192.168.0.107:5060;transport=udp>
Server: OpenSIPStack-1.1.6-132
Expires: 3600
Content-Length: 0

----------------5:47:02.795----------------
SEND: enc=0 767 Bytes to 70.42.73.140:5060:UDP (INVITE
sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107
INVITE sip:613@70.42.73.140 SIP/2.0
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: sip:613@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 4711 INVITE
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Contact: "300"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Type: application/sdp
Content-Length: 229

v=0
o=- 1182855278 1182855278 IN IP4 192.168.0.107
s=OSS RTP Session
c=IN IP4 192.168.0.107
t=0 0
m=audio 5000 RTP/AVP 101 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
----------------5:47:02.841----------------
SEND: enc=0 767 Bytes to 70.42.73.140:5060:UDP (INVITE
sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107
INVITE sip:613@70.42.73.140 SIP/2.0
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: sip:613@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 4711 INVITE
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Contact: "300"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Type: application/sdp
Content-Length: 229

v=0
o=- 1182855278 1182855278 IN IP4 192.168.0.107
s=OSS RTP Session
c=IN IP4 192.168.0.107
t=0 0
m=audio 5000 RTP/AVP 101 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
----------------5:47:02.928----------------
RCV: enc=0 563 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP
(SIP/2.0 407 Proxy Authentication Required)
SIP/2.0 407 Proxy Authentication Required
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: ;tag=e9128a6f04f918108bdfd184264b0e8e
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20
CSeq: 4711 INVITE
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Proxy-Authenticate: Digest realm=70.42.73.140,
nonce="ec52dfdc48854008a4e1f5b327b6a7ab",
opaque="4c7f21d239815332ea078702ccd86d83", algorithm=MD5
Content-Length: 0

----------------5:47:02.979----------------
SEND: enc=0 963 Bytes to 70.42.73.140:5060:UDP (INVITE
sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107
INVITE sip:613@70.42.73.140 SIP/2.0
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: sip:613@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bKe69f8f6f04f9181094daa2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 4712 INVITE
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Contact: "300"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Proxy-Authorization: Digest username="300", realm="70.42.73.140",
nonce="ec52dfdc48854008a4e1f5b327b6a7ab", uri="sip:613@70.42.73.140",
response="3606c0c2a3553f1be2de540d929d0965", algorithm=MD5
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Type: application/sdp
Content-Length: 229

v=0
o=- 1182855278 1182855278 IN IP4 192.168.0.107
s=OSS RTP Session
c=IN IP4 192.168.0.107
t=0 0
m=audio 5000 RTP/AVP 101 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
----------------5:47:02.944----------------
RCV: enc=0 563 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP
(SIP/2.0 407 Proxy Authentication Required)
SIP/2.0 407 Proxy Authentication Required
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: ;tag=e9128a6f04f918108bdfd184264b0e8e
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20
CSeq: 4711 INVITE
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Proxy-Authenticate: Digest realm=70.42.73.140,
nonce="ec52dfdc48854008a4e1f5b327b6a7ab",
opaque="4c7f21d239815332ea078702ccd86d83", algorithm=MD5
Content-Length: 0

----------------5:47:03.040----------------
SEND: enc=0 539 Bytes to 70.42.73.140:5060:UDP (ACK sip:613@70.42.73.140
SIP/2.0) Interface Address=192.168.0.107
ACK sip:613@70.42.73.140 SIP/2.0
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: ;tag=e9128a6f04f918108bdfd184264b0e8e
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 4711 ACK
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Contact: "300"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0

----------------5:47:03.029----------------
RCV: enc=0 563 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP
(SIP/2.0 407 Proxy Authentication Required)
SIP/2.0 407 Proxy Authentication Required
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: ;tag=e9128a6f04f918108bdfd184264b0e8e
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20
CSeq: 4711 INVITE
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Proxy-Authenticate: Digest realm=70.42.73.140,
nonce="ec52dfdc48854008a4e1f5b327b6a7ab",
opaque="4c7f21d239815332ea078702ccd86d83", algorithm=MD5
Content-Length: 0

----------------5:47:03.086----------------
SEND: enc=0 539 Bytes to 70.42.73.140:5060:UDP (ACK sip:613@70.42.73.140
SIP/2.0) Interface Address=192.168.0.107
ACK sip:613@70.42.73.140 SIP/2.0
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: ;tag=e9128a6f04f918108bdfd184264b0e8e
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 4711 ACK
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Contact: "300"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0

----------------5:47:03.129----------------
SEND: enc=0 539 Bytes to 70.42.73.140:5060:UDP (ACK sip:613@70.42.73.140
SIP/2.0) Interface Address=192.168.0.107
ACK sip:613@70.42.73.140 SIP/2.0
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: ;tag=e9128a6f04f918108bdfd184264b0e8e
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 4711 ACK
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Contact: "300"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0

----------------5:47:03.337----------------
SEND: enc=0 963 Bytes to 70.42.73.140:5060:UDP (INVITE
sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107
INVITE sip:613@70.42.73.140 SIP/2.0
From: 300 ;tag=24388f6f04f9181094daa2088e6bd734
To: sip:613@70.42.73.140
Via: SIP/2.0/UDP
192.168.0.107:5060;iid=1770;branch=z9hG4bKe69f8f6f04f9181094daa2088e6bd734;uas-addr=70.42.73.140;rport
CSeq: 4712 INVITE
Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734
Contact: "300"
User-Agent: OpenSIPStack-1.1.6-166
Max-Forwards: 10
Proxy-Authorization: Digest username="300", realm="70.42.73.140",
nonce="ec52dfdc48854008a4e1f5b327b6a7ab", uri="sip:613@70.42.73.140",
response="3606c0c2a3553f1be2de540d929d0965", algorithm=MD5
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Type: application/sdp
Content-Length: 229

v=0
o=- 1182855278 1182855278 IN IP4 192.168.0.107
s=OSS RTP Session
c=IN IP4 192.168.0.107
t=0 0
m=audio 5000 RTP/AVP 101 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

tomach wrote:

tomach

64 posts since
Jun 4, 2007

39. Jun 26, 2007 7:49 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hi!

Files that you asked and dates of their creation:

Softphone.h 2007.05.22
Softphone.cxx 2007.06.04
SoftphoneInterface.h 2007.05.31
SoftphoneInterface.cxx 2007.05.30

I belive that those are the newest one.

Guest

40. Jun 26, 2007 8:34 AM

in response to: tomach

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

Hi,

I''ve seen your logs and the From tag was definitely wrong but I can''t
reproduce this problem.

By the way, what app are you using? OSSPhone? Have you by any chance
coded something that sets the wrong domain before calling?

Regards,
Ilian

tomach wrote:

Hi!

Files that you asked and dates of their creation:

Softphone.h 2007.05.22
Softphone.cxx 2007.06.04
SoftphoneInterface.h 2007.05.31
SoftphoneInterface.cxx 2007.05.30

I belive that those are the newest one.

Guest

41. Jun 26, 2007 8:49 AM

in response to: tomach

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

I just committed a patch that might solve your problem. Make a fresh
CVS update. It is also important that you tell us if you made code
modifications.

tomach wrote:

Hi!

Files that you asked and dates of their creation:

Softphone.h 2007.05.22
Softphone.cxx 2007.06.04
SoftphoneInterface.h 2007.05.31
SoftphoneInterface.cxx 2007.05.30

I belive that those are the newest one.

Guest

42. Jun 26, 2007 8:49 AM

in response to: tomach

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

I just committed a patch that might solve your problem. Make a fresh
CVS update. It is also important that you tell us if you made code
modifications.

tomach wrote:

Hi!

Files that you asked and dates of their creation:

Softphone.h 2007.05.22
Softphone.cxx 2007.06.04
SoftphoneInterface.h 2007.05.31
SoftphoneInterface.cxx 2007.05.30

I belive that those are the newest one.

tomach

64 posts since
Jun 4, 2007

43. Jun 26, 2007 9:05 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

I was using both sample app and ossphone. I didnt code anything jsut run sample and try to make call. then i exmaine that everytime for FROM it took hostIP and it shoudl take domain and not my host ip.....

Im trying to find where to change it but i guess it will take time....Its really weird that you can not simulate this situation....

As i said even in SampleATL i just write

sip:8726@192.168.2.69

it sends invite where FROM is sip:192.168.2.45 where is should be 2.69....

weird that it only happend in my sample...

tomach

64 posts since
Jun 4, 2007

44. Jun 26, 2007 9:11 AM

in response to: Guest

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

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