[Asterisk-Users] Can't connect KPhone to asterisk
Regovich, Timothy
timothy_regovich at merck.com
Wed Feb 11 09:42:48 MST 2004
Not ACK'ing an invite can be problematic for the statemachine. Without the
ACK, the Dialog is not in acorrect state.
As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
is doing the same. There is no restriction that they must match. You can
change your offer in the ACK, or with a re-INVITE.
As for the immediate transmission : yeah, it does seem a little strange
doesn't it? But that is the way that I have seen almost all UAs work. The
implication is that your offer must be a valid, not a conditional offer :
when you say you accept GSM on port 8000, you better have a listener on 800
ready to go.
Tim
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Maciek Kaminski
Sent: Wednesday, February 11, 2004 11:39 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Can't connect KPhone to asterisk
Anyone managed to make KPhone work with Asterisk?
For me it looks as if KPhone does not ACK transactions, i.e.:
KPhone --INVITE--> Asterisk
Asterisk --Trying --> KPhone
Asterisk --OK --> KPhone
KPhone doest not acknowlege. Asterisk keeps resending OKs, KPhone
INVITES. Both timeouts.
By the way: KPhone offers PCMU, GSM, iLBC in INVITE, Asterisk answers
with PCMU and PCMA with doest not seem to be correct as it should answer
with subset of codecs offered(as far as I understood SIP RFC). Another
issue that bothers me is that Asterisk seems to start media transmission
as soon as it send OK not after it received ACK. Begining of
conversation may lost this way, isn't it?
Asterisk and KPhone logs below:
----------------------------------------------------------------------------
-----------------
Asterisk log:
----------------------------------------------------------------------------
-----------------
Sip read:
INVITE sip:700 at polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700 at polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
Call-ID: 1239913767 at 192.168.0.3
Subject: sip:maciejka at 192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski" <sip:maciejka at 192.168.0.3;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
11 headers, 9 lines
Using latest request as basis request
Sending to 192.168.0.3 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format GSM
Found description format iLBC
Capabilities: us - 12, them - 1030/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:4186 check_user: Setting NAT on
RTP to 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:5277 handle_request: Check for
res for maciejka
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:1128 find_user: Call from user
'maciejka' is 1 out of 0
Looking for 700 in default
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:3572 build_route: build_route:
Contact hop: "Maciek Kaminski" <sip:maciejka at 192.168.0.3;transport=udp>
list_route: hop: <sip:maciejka at 192.168.0.3;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
To: <sip:700 at polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767 at 192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700 at 192.168.0.2>
Content-Length: 0
to 192.168.0.3:5060
-- Executing Answer("SIP/maciejka-b4b6", "") in new stack
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
To: <sip:700 at polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767 at 192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700 at 192.168.0.2>
Content-Type: application/sdp
Content-Length: 153
v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
to 192.168.0.3:5060
-- Executing Festival("SIP/maciejka-b4b6", "Press 1 to heaven press
2 to go to hell press 3 to disconnect.") in new stack
== Parsing '/etc/asterisk/festival.conf': Found
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:318 festival_exec: Text
passed to festival server : Press 1 to heaven press 2 to go to hell
press 3 to disconnect.
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:395 festival_exec: Passing
text to festival...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:414 festival_exec: Passing
data to channel...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:424 festival_exec:
Festival WV command
Sip read:
INVITE sip:700 at polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700 at polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
Call-ID: 1239913767 at 192.168.0.3
Subject: sip:maciejka at 192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski" <sip:maciejka at 192.168.0.3;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
11 headers, 9 lines
Ignoring this request
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
To: <sip:700 at polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767 at 192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700 at 192.168.0.2>
Content-Type: application/sdp
Content-Length: 153
v=0
o=root 3363 3364 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
to 192.168.0.3:5060
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
To: <sip:700 at polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767 at 192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700 at 192.168.0.2>
Content-Type: application/sdp
Content-Length: 153
v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
to 192.168.0.3:5060
== Spawn extension (default, 700, 2) exited non-zero on
'SIP/maciejka-b4b6'
Feb 11 17:12:37 DEBUG[180236]: chan_sip.c:1207 sip_hangup:
find_user(maciejka) - decrement inUse counter
set_destination: Parsing <sip:maciejka at 192.168.0.3;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.0.3, port 5060
Reliably Transmitting:
BYE sip:maciejka at 192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK77cfdb1c
From: <sip:700 at polimorfia>;tag=as3b0a9ff0
To: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
Contact: <sip:700 at 192.168.0.2>
Call-ID: 1239913767 at 192.168.0.3
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.3:5060
----------------------------------------------------------------------------
-----------------
KPhone log:
----------------------------------------------------------------------------
-----------------
SipClient: Sending: 17:18:40.048
--------------------------------
INVITE sip:700 at polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700 at polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
Call-ID: 1239913767 at 192.168.0.3
Subject: sip:maciejka at 192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski" <sip:maciejka at 192.168.0.3;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
SipClient: Sending to '192.168.0.2:5060'
SipClient: Receiving message...
SipClient: Received: 17:18:40.168
---------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
To: <sip:700 at polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767 at 192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700 at 192.168.0.2>
Content-Length: 0
SipClient: Receiving message...
SipClient: Received: 17:18:40.182
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
To: <sip:700 at polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767 at 192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700 at 192.168.0.2>
Content-Type: application/sdp
Content-Length: 153
v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
SipTransaction: Retransmit 1 (500)
SipClient: Sending: 17:18:40.551
--------------------------------
INVITE sip:700 at polimorfia SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: <sip:700 at polimorfia>
Content-Type: application/sdp
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
Call-ID: 1239913767 at 192.168.0.3
Subject: sip:maciejka at 192.168.0.3
Content-Length: 183
User-Agent: kphone/4.0
Contact: "Maciek Kaminski" <sip:maciejka at 192.168.0.3;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
SipClient: Receiving message...
SipClient: Received: 17:18:40.568
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: "Maciek Kaminski" <sip:maciejka at 192.168.0.3>;tag=B62B188
To: <sip:700 at polimorfia>;tag=as3b0a9ff0
Call-ID: 1239913767 at 192.168.0.3
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:700 at 192.168.0.2>
Content-Type: application/sdp
Content-Length: 153
v=0
o=root 3363 3364 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
SipClient: Receiving message...
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