[Asterisk-Users] chan_oh323.c:2501 ast_oh323_new: Internal channel
initialization failed. Bad binary?
Kamran Ahmad
p_kami at yahoo.com
Wed Mar 16 02:07:49 MST 2005
hello
i try to call from sip phone on asteris to open phone
on GnuGK.
can any one tell me why it is saying
chan_oh323.c:2501 ast_oh323_new: Internal channel
initialization failed. Bad binary?
Mar 16 13:28:46 WARNING[5963]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
structure 4.
Mar 16 13:28:46 NOTICE[5963]: app_dial.c:749
dial_exec: Unable to create channel of type 'OH323'
We're at 192.168.0.203 port 17456
------------------------------------------------
Sip read:
INVITE sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From:<sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
Contact: <sip:2000 at 192.168.0.153>
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="6ebe9c68",uri="sip:192.168.0.203",response="7027ef8069a0ef7a5f8089fda2fc0e87"
v=0
o=sibtay 2890844 842807 IN IP4 192.168.0.153
s=SDP Seminar
c=IN IP4 192.168.0.153
t=0 0
m=audio 13064 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 192.168.0.153 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.153:13064
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 3218888 in default
list_route: hop: <sip:2000 at 192.168.0.153>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Length: 0
to 192.168.0.153:5060
Mar 16 13:28:34 ERROR[5963]: chan_oh323.c:2501
ast_oh323_new: Internal channel initialization failed.
Bad binary?
Mar 16 13:28:34 WARNING[5963]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
structure 3.
Mar 16 13:28:34 NOTICE[5963]: app_dial.c:749
dial_exec: Unable to create channel of type 'OH323'
*CLI>
*CLI>
Sip read:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>
Call-ID: 52 at 192.168.0.153
CSeq: 22 INFO
Contact: <sip:2000 at 192.168.0.153>
Content-Type: application/dtmf-relay
Content-Length: 26
Signal= 8
Duration= 160
9 headers, 4 lines
Receiving DTMF!
* DTMF received: '8'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 22 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Length: 0
to 192.168.0.153:5060
Sip read:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>
Call-ID: 52 at 192.168.0.153
CSeq: 23 INFO
Contact: <sip:2000 at 192.168.0.153>
Content-Type: application/dtmf-relay
Content-Length: 26
Signal= 8
Duration= 160
9 headers, 4 lines
Receiving DTMF!
* DTMF received: '8'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 23 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Length: 0
to 192.168.0.153:5060
Sip read:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>
Call-ID: 52 at 192.168.0.153
CSeq: 24 INFO
Contact: <sip:2000 at 192.168.0.153>
Content-Type: application/dtmf-relay
Content-Length: 26
Signal= 8
Duration= 160
9 headers, 4 lines
Receiving DTMF!
* DTMF received: '8'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 24 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Length: 0
to 192.168.0.153:5060
Sip read:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>
Call-ID: 52 at 192.168.0.153
CSeq: 25 INFO
Contact: <sip:2000 at 192.168.0.153>
Content-Type: application/dtmf-relay
Content-Length: 26
Signal= 8
Duration= 160
9 headers, 4 lines
Receiving DTMF!
* DTMF received: '8'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:172.16.0.32>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 25 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Length: 0
to 192.168.0.153:5060
Mar 16 13:28:46 ERROR[5963]: chan_oh323.c:2501
ast_oh323_new: Internal channel initialization failed.
Bad binary?
Mar 16 13:28:46 WARNING[5963]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
structure 4.
Mar 16 13:28:46 NOTICE[5963]: app_dial.c:749
dial_exec: Unable to create channel of type 'OH323'
We're at 192.168.0.203 port 17456
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
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 5963 5963 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 17456 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 192.168.0.153:5060
Sip read:
ACK sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq: 21 ACK
6 headers, 0 lines
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 5963 5963 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 17456 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 192.168.0.153:5060
Sip read:
ACK sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq: 21 ACK
6 headers, 0 lines
Retransmitting #2 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 5963 5963 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 17456 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 192.168.0.153:5060
Sip read:
ACK sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq: 21 ACK
6 headers, 0 lines
Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 5963 5963 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 17456 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 192.168.0.153:5060
Sip read:
ACK sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq: 21 ACK
6 headers, 0 lines
Retransmitting #4 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 5963 5963 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 17456 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 192.168.0.153:5060
Sip read:
ACK sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq: 21 ACK
6 headers, 0 lines
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as61b12c41
Call-ID: 52 at 192.168.0.153
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3218888 at 192.168.0.203>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 5963 5963 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 17456 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 192.168.0.153:5060
Sip read:
ACK sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: <sip:2000 at 192.168.0.203>
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq: 21 ACK
6 headers, 0 lines
Mar 16 13:29:02 WARNING[5963]: chan_sip.c:694
retrans_pkt: Maximum retries exceeded on call
52 at 192.168.0.153 for seqno 21 (Non-critical Response)
Destroying call '52 at 192.168.0.153'
Sip read:
BYE sip:3218888 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>
Call-ID: 52 at 192.168.0.153
CSeq:21
Contact= <sip:2000 at 192.168.0.153>
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
13 headers, 0 lines
Sending to 192.168.0.153 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: <sip:2000 at 192.168.0.203>;
To: <sip:3218888 at 192.168.0.203>;tag=as31bcfc3e
Call-ID: 52 at 192.168.0.153
CSeq: 21
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 192.168.0.153:5060
Destroying call '52 at 192.168.0.153'
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