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