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<DIV><FONT face=Arial size=2>Hi all,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> I am trying to use a "Siemens optiPoint 300"
IPPhone (H.323 only) with Asterisk (1.0-RC2).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> So far I have been using the H.323 channel
included in the tarball (Nufone ?).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> I encountered a strange behaviour when I try
to make a call from the IPPhone to my Asterisk box :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> =====> here is the H.323
configuration for the incoming calls (192.168.1.50 is the IP of the Siemens
phone</FONT><FONT face=Arial size=2>) :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[Damien]<BR>type=user<BR>host=192.168.1.50<BR>context=incoming<BR></FONT><FONT
face=Arial size=2></FONT></DIV>
<DIV><FONT face=Arial size=2> =====> the incoming context
has a single extension :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[incoming]</FONT></DIV>
<DIV><FONT face=Arial size=2>exten => 666,1,Playback(demo-echotest)<BR>exten
=>
666,2,Echo
<BR>exten => 666,3,Playback(demo-echodone) <BR></FONT></DIV>
<DIV><FONT face=Arial size=2> =====> the IPPhone is configured to use
the Asterisk box as a H.323 gateway (system type = "Gateway" / IP address of the
gateway = IP address of the Asterisk box)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> =====> when I dial "666" on the IPPhone
Asterisk seems to answer the call then 2 seconds later it hangs up
:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>***** DEBUG MESSAGES DURING THE CALL
*****</FONT></DIV>
<DIV><FONT face=Arial size=2>== New H.323 Connection
created.<BR> -- Received SETUP
message<BR> -- Setting up Call<BR>
-- Call token:
[ip$192.168.1.50:1257/5625]<BR> --
Calling party name: []<BR> --
Calling party number: [987654321]<BR>
-- Called party name:
[666]<BR> -- Called party
number: [666]<BR>Urgent handler<BR>Aug 30 11:48:32 DEBUG[56142768]:
pbx.c:1255 pbx_extension_helper: Launching 'Playback'<BR>Aug 30 11:48:32
DEBUG[56142768]: channel.c:1666 ast_set_write_format: Set channel
H323/ip$192.168.1.50:1257/5625 to write format
GSM<BR> -- Received RELEASE COMPLETE
message...<BR> -- Sending RELEASE
COMPLETE<BR>
1:32.765
H245:8a174a0
h323.cxx(3195) H245 Read error: Interrupted system
call<BR>
1:32.781
H323 Cleaner
h323.cxx(1542) H323 Connection ip$192.168.1.50:1257/5625
terminated.<BR> -- 987654321, 987654321 [192.168.1.50] has cleared the
call<BR>Aug 30 11:48:35 DEBUG[56142768]: channel.c:1666 ast_set_write_format:
Set channel H323/ip$192.168.1.50:1257/5625 to write format ALAW<BR>Aug 30
11:48:35 DEBUG[56142768]: pbx.c:1827 ast_pbx_run: Spawn extension
(incoming,666,1) exited non-zero on 'H323/ip$192.168.1.50:1257/5625'<BR>Aug 30
11:48:35 DEBUG[56142768]: channel.c:733 ast_hangup: Hanging up channel
'H323/ip$192.168.1.50:1257/5625'<BR>Aug 30 11:48:35 DEBUG[56142768]:
chan_h323.c:531 oh323_hangup:
oh323_hangup(H323/ip$192.168.1.50:1257/5625)<BR>
== H.323 Connection deleted.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> =====> if I had a "Wait 1" in front the
extension it works :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[incoming]<BR>exten => 666,1,Wait,1<BR>exten
=> 666,2,Playback(demo-echotest)<BR>exten => 666,3,Echo<BR>exten =>
666,4,Playback(demo-echodone)<BR></FONT></DIV>
<DIV>
<DIV><FONT face=Arial size=2>***** DEBUG MESSAGES DURING THE CALL
*****</FONT></DIV></DIV>
<DIV><FONT face=Arial size=2>== New H.323 Connection
created.<BR> -- Received SETUP
message<BR>Urgent handler<BR> -- Setting up
Call<BR> -- Call token:
[ip$192.168.1.50:1260/5626]<BR> --
Calling party name: []<BR> --
Calling party number: [987654321]<BR>
-- Called party name:
[666]<BR> -- Called party
number: [666]<BR>Urgent handler<BR>Aug 30 11:53:34 DEBUG[114731952]:
pbx.c:1255 pbx_extension_helper: Launching 'Wait'<BR>Aug 30 11:53:35
DEBUG[114731952]: pbx.c:1255 pbx_extension_helper: Launching
'Playback'<BR> =*= In
CreateRealTimeLogicalChannel for call
5626<BR>
-- externalIpAddress:
192.168.1.201<BR>
-- externalPort:
15508<BR>
-- SessionID:
1<BR>
-- Direction: IsTransmitter<BR>
-- Started logical channel: sending
G.711-ALaw-64k{sw}<BR>
-- channelsOpen = 1<BR> -- Connection
Established with "987654321, 987654321 [192.168.1.50]"<BR>Aug 30 11:53:35
DEBUG[114731952]: channel.c:1666 ast_set_write_format: Set channel
H323/ip$192.168.1.50:1260/5626 to write format
GSM<BR> =*= In
CreateRealTimeLogicalChannel for call
5626<BR>
-- externalIpAddress:
192.168.1.201<BR>
-- externalPort:
15508<BR>
-- SessionID:
1<BR>
-- Direction: IsReceiver<BR> --
Started logical channel: receiving
G.711-ALaw-64k{sw}<BR>
-- channelsOpen = 2<BR>Aug 30 11:53:36 DEBUG[114731952]: rtp.c:1156
ast_rtp_write: Ooh, format changed from UNKN to ALAW<BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> Any idea about this "H245
Read error: Interrupted system call" that appears in the debug
messages ???</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks, Damien.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>BTW, the H.323 channel has been compiled with the
recommended PWLib 1.5.2 and OpenH323 1.12.2.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV></BODY></HTML>