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

Fri Aug 24 10:37:20 CDT 2007

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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