hello
i want to rout my calls to h.323. i have registered my
asterisk with GnuGatekeeper. but it is not routing my
call to h.323 channel. he is saying Internal channel
initialization failed. Bad binary?
can any one check my settings what is problem here
thanks in advance
kamran
exten=>_321XXXX,1,Dial(OH323/${EXTEN}@192.168.0.153:1719,30,r)
------------------------------------------------------
*CLI> -- Registered with gatekeeper 'GNU
Gatekeeper@192.168.0.153'.
Sip read:
INVITE sip:3214567@192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From:<sip:2000@192.168.0.203>;
To: <sip:3214567@192.168.0.203>
Call-ID: 51@192.168.0.117
CSeq: 20 INVITE
Contact: <sip:2000@192.168.0.117>
Max-Forwards: 5
User-Agent:SKYPHONE/1.03
Subject: hello
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS,
REFER,SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:180
v=0
o=sibtay 2890844 842807 IN IP4 192.168.0.117
s=SDP Seminar
c=IN IP4 192.168.0.117
t=0 0
m=audio 13044 RTP/AVP 0 101
a=rtpmap:101 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:96 0-11,16
14 headers, 10 lines
Using latest request as basis request
Sending to 192.168.0.117 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.117:13044
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer
- audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4
(ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1
(g723), combined - 0x1 (g723)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From: <sip:2000@192.168.0.203>;
To: <sip:3214567@192.168.0.203>;tag=as6d7474f0
Call-ID: 51@192.168.0.117
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3214567@192.168.0.203>
Proxy-Authenticate: Digest realm="asterisk",
nonce="1947602b"
Content-Length: 0
to 192.168.0.117:5060
Scheduling destruction of call '51@192.168.0.117' in
15000 ms
Found user '2000'
Sip read:
ACK sip:3214567@192.168.0.203 SIP/2.0
Content-Length: 0
Call-ID: 51@192.168.0.117
Max-Forwards: 70
CSeq: 20 ACK
From: <sip:2000@192.168.0.203>
To: <sip:3214567@192.168.0.203>
Via: SIP/2.0/UDP 192.168.0.117
8 headers, 0 lines
Sip read:
INVITE sip:3214567@192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From:<sip:2000@192.168.0.203>;
To: <sip:3214567@192.168.0.203>
Call-ID: 51@192.168.0.117
CSeq: 21 INVITE
Contact: <sip:2000@192.168.0.117>
Max-Forwards: 5
User-Agent:SKYPHONE/1.03
Subject: hello
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS,
REFER,SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:180
Proxy-Authorization: Digest
username="2000",realm="asterisk",nonce="1947602b",uri="sip:192.168.0.203",response="11ef2adb8d7b567c2c5d7e7c3a1aea61"
v=0
o=sibtay 2890844 842807 IN IP4 192.168.0.117
s=SDP Seminar
c=IN IP4 192.168.0.117
t=0 0
m=audio 13044 RTP/AVP 0 101
a=rtpmap:101 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:96 0-11,16
15 headers, 11 lines
Using latest request as basis request
Sending to 192.168.0.117 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.117:13044
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer
- audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4
(ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1
(g723), combined - 0x1 (g723)
Found user '2000'
Looking for 3214567 in default
list_route: hop: <sip:2000@192.168.0.117>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From: <sip:2000@192.168.0.203>;
To: <sip:3214567@192.168.0.203>;tag=as2d54adea
Call-ID: 51@192.168.0.117
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3214567@192.168.0.203>
Content-Length: 0
to 192.168.0.117:5060
Mar 15 13:43:23 ERROR[4401]: chan_oh323.c:2501
ast_oh323_new: Internal channel initialization failed.
Bad binary?
Mar 15 13:43:23 WARNING[4401]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
structure 0.
Mar 15 13:43:23 NOTICE[4401]: app_dial.c:749
dial_exec: Unable to create channel of type 'OH323'
Mar 15 13:43:33 NOTICE[4401]: rtp.c:430 ast_rtp_read:
RTP: Received packet with bad UDP checksum
We're at 192.168.0.203 port 10418
Answering with preferred capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From: <sip:2000@192.168.0.203>;
To: <sip:3214567@192.168.0.203>;tag=as2d54adea
Call-ID: 51@192.168.0.117
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3214567@192.168.0.203>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 4401 4401 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 10418 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 192.168.0.117:5060
Sip read:
ACK sip:3214567@192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.117
From: <sip:2000@192.168.0.203>
To: <sip:3214567@192.168.0.203>
Call-ID: 51@192.168.0.117
CSeq: 21 ACK
6 headers, 0 lines
set_destination: Parsing <sip:2000@192.168.0.117> for
address/port to send to
set_destination: set destination to 192.168.0.117,
port 5060
Reliably Transmitting:
BYE sip:2000@192.168.0.117 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.203:5060;branch=z9hG4bK26652990;rport
From: <sip:3214567@192.168.0.203>;tag=as2d54adea
To: <sip:2000@192.168.0.203>;
Contact: <sip:3214567@192.168.0.203>
Call-ID: 51@192.168.0.117
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.117:5060
Sip read:
SIP/2.0 200 OK
From:<tag>;
To: <Contact:>
Call-ID: 51@192.168.0.117
CSeq: 102 BYE
Via: SIP/2.0/UDP
192.168.0.203:5060;branch=z9hG4bK26652990
Content-Length: 0
User-Agent: SKYPHONE/1.03
Contact: <2000@192.168.0.117>
9 headers, 0 lines
Message is BYE
Destroying call '51@192.168.0.117'
*CLI>
*CLI> show chann
channel channels
*CLI> show channels
Channel (Context Extension Pri )
State Appl. Data
0 active channel(s)
*CLI>
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