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Kutman.DK at forces.gc.ca

2007-Aug-24 14:41 UTC

[asterisk-users] Can't create audio conversation between softphones through Asterisk

Hello,

I have two user machines, each with a jain-sip-applet-phone installed on it.  I
use the following process to try to make a call:

1.  Register each phone with the Asterisk server (working).
2.  Add a contact in each phone which is the other user. (Get a "489 Bad
Event" SIP error shown below in red)

202 at 192.168.1.252 has been added to your contacts.
null
send request:
SUBSCRIBE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251
CSeq: 1 SUBSCRIBE
From: <sip:201 at 192.168.1.251>;tag=8505
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Length: 0


<message
from="192.168.1.251:8386" 
to="192.168.1.10:5060" 
time="1187721756281" 
isSender="true" 
transactionId="z9hg4bk361290cad5885dbc4a03b5951cc85585" 
callId="59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251" 
firstLine="SUBSCRIBE sip:202 at 192.168.1.252;transport=udp SIP/2.0" 
debugLine="0" ><![CDATA[SUBSCRIBE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251
CSeq: 1 SUBSCRIBE
From: <sip:201 at 192.168.1.251>;tag=8505
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Length: 0

]]>
</message>

<message
from="192.168.1.10:5060" 
to="192.168.1.251:8386" 
time="1187721756281" 
isSender="false" 
statusMessage="normal processing" 
transactionId="z9hg4bk361290cad5885dbc4a03b5951cc85585" 
firstLine="SIP/2.0 489 Bad Event" 
callId="59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251" 
debugLine="0" ><![CDATA[SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585;received=192.168.1.251
From: <sip:201 at 192.168.1.251>;tag=8505
To: <sip:202 at 192.168.1.252>;tag=as2cf724e9
Call-ID: 59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0

3.  Try to call that contact to create an audio conversation.(Get a "488
Not Acceptable Here" SIP error shown below in blue)

Get chat session: 202 at 192.168.1.252
Chat Session added: 202 at
192.168.1.252:gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,hidden,layout=java.awt.BorderLayout,title=In
conversation with 202 at
192.168.1.252,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,104x1,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]
5
4
3
0
send request:
INVITE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251
CSeq: 1 INVITE
From: <sip:201 at 192.168.1.251>;tag=2085
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Type: application/sdp
Content-Length: 114

v=0
o=201 908031 909400 IN IP4 192.168.1.251
s=-
c=IN IPV4 192.168.1.251
t=0 0
m=audio 2448 RTP/AVP 5 4 3 0

<message
from="192.168.1.251:8386" 
to="192.168.1.10:5060" 
time="1187721758593" 
isSender="true" 
transactionId="z9hg4bk583c7ea7c1a8da8576583356f821c9c2" 
callId="8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251" 
firstLine="INVITE sip:202 at 192.168.1.252;transport=udp SIP/2.0" 
debugLine="0" ><![CDATA[INVITE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251
CSeq: 1 INVITE
From: <sip:201 at 192.168.1.251>;tag=2085
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Type: application/sdp
Content-Length: 114

]]>
</message>

<message
from="192.168.1.10:5060" 
to="192.168.1.251:8386" 
time="1187721758609" 
isSender="false" 
statusMessage="normal processing" 
transactionId="z9hg4bk583c7ea7c1a8da8576583356f821c9c2" 
firstLine="SIP/2.0 488 Not acceptable here" 
callId="8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251" 
debugLine="0" ><![CDATA[SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2;received=192.168.1.251
From: <sip:201 at 192.168.1.251>;tag=2085
To: <sip:202 at 192.168.1.252>;tag=as2f851644
Call-ID: 8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0

Has anyone ever tried using these Jain-sip-applet-phones and got them to work? 
I have read up on these errors, and it looks like the 489 error doesn't like
the SUBSCRIBE request, while the 488 error doesn't seem to accept the INVITE
request made.  I am not sure if this is a problem with Asterisk, incompatibility
between Asterisk and the phones, or just the phones.  Any thoughts that may help
me resolve these issues would be greatly appreciated.

Thanks very much,

Denis

Kutman.DK at forces.gc.ca

2007-Aug-24 15:37 UTC

head link

[asterisk-users] Can't create audio conversation between softphonesthrough Asterisk

This is the "full" log that I get after my trial run:

Aug 24 14:15:51 VERBOSE[3710] logger.c:     -- Registered SIP '202' at
192.168.1.250 port 9810 expires 120
Aug 24 14:15:52 VERBOSE[3710] logger.c:     -- Registered SIP '201' at
192.168.1.251 port 8529 expires 120
Aug 24 14:15:55 NOTICE[3710] chan_sip.c: Peer '202' is now UNREACHABLE! 
Last qualify: 0
Aug 24 14:15:56 NOTICE[3710] chan_sip.c: Peer '201' is now UNREACHABLE! 
Last qualify: 0
Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call
'f27bb8c878b2d80cc886f9d223c25631 at 192.168.1.250'
Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call
'c627ed1d462b8456370f92b8e472880b at 192.168.1.251'
Aug 24 14:16:10 DEBUG[3710] chan_sip.c: Setting NAT on RTP to 0
Aug 24 14:16:10 WARNING[3710] chan_sip.c: Invalid host in c= line, 'IN IPV4
192.168.1.251'
Aug 24 14:16:10 DEBUG[3710] chan_sip.c: SIP message could not be handled, bad
request: b475318241b3dca93128681e6f079093
192.168.1.251    

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of
Kutman.DK at forces.gc.ca
Sent: Friday, August 24, 2007 10:41 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Can't create audio conversation between
softphonesthrough Asterisk


Hello,

I have two user machines, each with a jain-sip-applet-phone installed on it.  I
use the following process to try to make a call:

1.  Register each phone with the Asterisk server (working).
2.  Add a contact in each phone which is the other user. (Get a "489 Bad
Event" SIP error shown below in red)

202 at 192.168.1.252 has been added to your contacts.
null
send request:
SUBSCRIBE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251
CSeq: 1 SUBSCRIBE
From: <sip:201 at 192.168.1.251>;tag=8505
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Length: 0


<message
from="192.168.1.251:8386" 
to="192.168.1.10:5060" 
time="1187721756281" 
isSender="true" 
transactionId="z9hg4bk361290cad5885dbc4a03b5951cc85585" 
callId="59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251" 
firstLine="SUBSCRIBE sip:202 at 192.168.1.252;transport=udp SIP/2.0" 
debugLine="0" ><![CDATA[SUBSCRIBE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251
CSeq: 1 SUBSCRIBE
From: <sip:201 at 192.168.1.251>;tag=8505
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Length: 0

]]>
</message>

<message
from="192.168.1.10:5060" 
to="192.168.1.251:8386" 
time="1187721756281" 
isSender="false" 
statusMessage="normal processing" 
transactionId="z9hg4bk361290cad5885dbc4a03b5951cc85585" 
firstLine="SIP/2.0 489 Bad Event" 
callId="59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251" 
debugLine="0" ><![CDATA[SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585;received=192.168.1.251
From: <sip:201 at 192.168.1.251>;tag=8505
To: <sip:202 at 192.168.1.252>;tag=as2cf724e9
Call-ID: 59f9fb5dd6b9fa775e38c6f31671bbc4 at 192.168.1.251
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0

3.  Try to call that contact to create an audio conversation.(Get a "488
Not Acceptable Here" SIP error shown below in blue)

Get chat session: 202 at 192.168.1.252
Chat Session added: 202 at
192.168.1.252:gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,hidden,layout=java.awt.BorderLayout,title=In
conversation with 202 at
192.168.1.252,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,104x1,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]
5
4
3
0
send request:
INVITE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251
CSeq: 1 INVITE
From: <sip:201 at 192.168.1.251>;tag=2085
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Type: application/sdp
Content-Length: 114

v=0
o=201 908031 909400 IN IP4 192.168.1.251
s=-
c=IN IPV4 192.168.1.251
t=0 0
m=audio 2448 RTP/AVP 5 4 3 0

<message
from="192.168.1.251:8386" 
to="192.168.1.10:5060" 
time="1187721758593" 
isSender="true" 
transactionId="z9hg4bk583c7ea7c1a8da8576583356f821c9c2" 
callId="8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251" 
firstLine="INVITE sip:202 at 192.168.1.252;transport=udp SIP/2.0" 
debugLine="0" ><![CDATA[INVITE sip:202 at 192.168.1.252;transport=udp SIP/2.0
Call-ID: 8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251
CSeq: 1 INVITE
From: <sip:201 at 192.168.1.251>;tag=2085
To: <sip:202 at 192.168.1.252>
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2
Max-Forwards: 2
Contact: <sip:201 at 192.168.1.251:8386;transport=udp>
Content-Type: application/sdp
Content-Length: 114

]]>
</message>

<message
from="192.168.1.10:5060" 
to="192.168.1.251:8386" 
time="1187721758609" 
isSender="false" 
statusMessage="normal processing" 
transactionId="z9hg4bk583c7ea7c1a8da8576583356f821c9c2" 
firstLine="SIP/2.0 488 Not acceptable here" 
callId="8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251" 
debugLine="0" ><![CDATA[SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2;received=192.168.1.251
From: <sip:201 at 192.168.1.251>;tag=2085
To: <sip:202 at 192.168.1.252>;tag=as2f851644
Call-ID: 8d6d2f3a9e5bd7f4a3aca5f729809b3c at 192.168.1.251
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0

Has anyone ever tried using these Jain-sip-applet-phones and got them to work? 
I have read up on these errors, and it looks like the 489 error doesn't like
the SUBSCRIBE request, while the 488 error doesn't seem to accept the INVITE
request made.  I am not sure if this is a problem with Asterisk, incompatibility
between Asterisk and the phones, or just the phones.  Any thoughts that may help
me resolve these issues would be greatly appreciated.

Thanks very much,

Denis


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