Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
----------------------------------------------------------------------
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: [email protected]
Contact: <sip:82.101.62.99:5060>
Content-Type: application/sdp
CSeq: 103 INVITE
From: "**********" <sip:*******[email protected]>;tag=as70e84199
Record-Route:
<sip:82.101.62.115;lr;r2=on;ftag=as70e84199>,<sip:82.101.63.5;lr;r2=on;ftag=as70e84199>
Server: Cirpack/v4.41b (gw_sip)
To: <sip:0031*****[email protected]>;tag=00-08168-044b6f36-245cd72c7
Via: SIP/2.0/UDP
***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92
Content-Length: 182
v=0
o=cp10 125193221174 125193221174 IN IP4 82.101.62.66
s=SIP Call
c=IN IP4 194.109.8.2
t=0 0
m=audio 36984 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 194.109.8.2:36984
Found audio description format PCMA for ID 8
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8
(alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.109.8.2:36984
-- SIP/*********-089ca9b8 is ringing
-- SIP/*********-089ca9b8 is making progress passing it to
IAX2/2104-2287
Scheduling destruction of SIP dialog
'[email protected]' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 82.101.62.99:5060:
CANCEL sip:0031******[email protected] SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport
From: "**********" <sip:******[email protected]>;tag=as70e84199
To: <sip:0031******[email protected]>
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
----------------------------------------------------------------------
However when I dial exactly the same from VoipBuster, I see this instead:
----------------------------------------------------------------------
<--- SIP read from 77.72.169.129:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport
From: "*********" <sip:******[email protected]>;tag=as1374705a
To: <sip:0031******[email protected]>;tag=120113ac4a54a269af9e2c
Contact: sip:0031******[email protected]:5060
Call-ID: [email protected]
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 162
v=0
o=********* 1251932194 1251932194 IN IP4 194.221.62.33
s=SIP Call
c=IN IP4 194.221.62.33
t=0 0
m=audio 8958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 194.221.62.33:8958
Found audio description format PCMU for ID 0
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.221.62.33:8958
-- SIP/********-089dc538 is making progress passing it to IAX2/2104-8077
== Connect attempt from '127.0.0.1' unable to authenticate
Scheduling destruction of SIP dialog
'[email protected]' in 6400 ms
(Method: INVITE)
Reliably Transmitting (NAT) to 77.72.169.129:5060:
CANCEL sip:0031******[email protected] SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport
From: "**********" <sip:*******[email protected]>;tag=as1374705a
To: <sip:0031******[email protected]>
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
----------------------------------------------------------------------
As you can see, there are different packets being sent, and in the 2nd
case, there is no "is ringing" message, which is rather irritating...
Any suggestions would be appreciated...
TIA
--
FP
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