[Asterisk-Users] Can't connect KPhone to asterisk

Wed Feb 11 09:38:46 MST 2004

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...