Re: [ATLSIP] ATLSIP and .Net

[Jonas G. <jon...@gm...>]

Ignore the event part of my previous message. The events work fine.

I still haven't got makeCall to work. Here is a log:


----------------7:49:55.642----------------
SEND: XOR=0 434 Bytes to 192.168.0.58:5060:UDP (REGISTER
sip:192.168.0.58 SIP/2.0) Interface Address=
REGISTER sip:192.168.0.58 SIP/2.0
From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1
To: sip:jo...@th...
Via: SIP/2.0/UDP
192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport
CSeq: 1 REGISTER
Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1
Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp>
Expires: 3600
Max-Forwards: 70
Content-Length: 0


----------------7:49:55.799----------------
RCV: XOR=0 627 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP
(SIP/2.0 200 OK)
SIP/2.0 200 OK
From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1
To:  <sip:jo...@th...>;tag=KUD9S70vH8X7a
Via: SIP/2.0/UDP
192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061
CSeq: 1 REGISTER
Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1
Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp>;expires=3600
User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, precondition, timer
Content-Length: 0


----------------7:49:57.641----------------
SEND: XOR=0 674 Bytes to 192.168.0.58:5060:UDP (INVITE
sip:83...@th... SIP/2.0) Interface Address=192.168.0.58
INVITE sip:83...@th... SIP/2.0
From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1
To: sip:83...@th...
Via: SIP/2.0/UDP
192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport
CSeq: 4711 INVITE
Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1
Contact: "jonas" <sip:jonas@192.168.0.58:5061>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 230

v=0
o=- 1193852436 1193852436 IN IP4 192.168.0.58
s=OSS RTP Session
c=IN IP4 192.168.0.58
t=0 0
m=audio 5000 RTP/AVP 101 106 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:106 iLBC/8000
a=rtpmap:3 GSM/8000



----------------7:49:57.648----------------
RCV: XOR=0 381 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP
(SIP/2.0 100 Trying)
SIP/2.0 100 Trying
From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1
To:  <sip:83...@th...>
Via: SIP/2.0/UDP
192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061
CSeq: 4711 INVITE
Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1
User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia
Content-Length: 0


----------------7:49:57.659----------------
RCV: XOR=0 594 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP
(SIP/2.0 488 Not Acceptable Here)
SIP/2.0 488 Not Acceptable Here
From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1
To:  <sip:83...@th...>;tag=m461U2H0eHmtp
Via: SIP/2.0/UDP
192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061
CSeq: 4711 INVITE
Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1
User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, precondition, timer
Content-Length: 0


----------------7:49:57.663----------------
SEND: XOR=0 422 Bytes to 192.168.0.58:5060:UDP (ACK
sip:83...@th... SIP/2.0) Interface Address=192.168.0.58
ACK sip:83...@th... SIP/2.0
From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1
To:  <sip:83...@th...>;tag=m461U2H0eHmtp
Via: SIP/2.0/UDP
192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport
CSeq: 4711 ACK
Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1
Contact: "jonas" <sip:jonas@192.168.0.58:5061>
Max-Forwards: 70
Content-Length: 0




On 10/31/07, Jonas Gauffin <jon...@gm...> wrote:
> Hello
>
> I've downloaded ATLsip and is testing it with C#.
> I'm using FreeSWITCH as sip pbx.
>
> I can register OK, but I do not receive any events to any of the
> handler methods. Any suggestions on why?
>
> What should the SIP Uri look like when making calls? I've tried some
> different variants, but FreeSWITCH returns  INCOMPATIBLE_DESTINATION
> on the INVITE.
>
> Thanks,
>   Jonas
>