Kamran Ahmad
2005-Mar-16 02:07 UTC
[Asterisk-Users] chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary?
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@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153;branch=z9hG4bK2038176231 From:<sip:2000@192.168.0.203>; To: <sip:3218888@192.168.0.203> Call-ID: 52@192.168.0.153 CSeq: 21 INVITE Contact: <sip:2000@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@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203> To: <sip:172.16.0.32> Call-ID: 52@192.168.0.153 CSeq: 22 INFO Contact: <sip:2000@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@192.168.0.203> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 22 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203> To: <sip:172.16.0.32> Call-ID: 52@192.168.0.153 CSeq: 23 INFO Contact: <sip:2000@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@192.168.0.203> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 23 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203> To: <sip:172.16.0.32> Call-ID: 52@192.168.0.153 CSeq: 24 INFO Contact: <sip:2000@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@192.168.0.203> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 24 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203> To: <sip:172.16.0.32> Call-ID: 52@192.168.0.153 CSeq: 25 INFO Contact: <sip:2000@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@192.168.0.203> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 25 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:2000@192.168.0.203> To: <sip:3218888@192.168.0.203> Call-ID: 52@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:2000@192.168.0.203> To: <sip:3218888@192.168.0.203> Call-ID: 52@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:2000@192.168.0.203> To: <sip:3218888@192.168.0.203> Call-ID: 52@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:2000@192.168.0.203> To: <sip:3218888@192.168.0.203> Call-ID: 52@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:2000@192.168.0.203> To: <sip:3218888@192.168.0.203> Call-ID: 52@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as61b12c41 Call-ID: 52@192.168.0.153 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3218888@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@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:2000@192.168.0.203> To: <sip:3218888@192.168.0.203> Call-ID: 52@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@192.168.0.153 for seqno 21 (Non-critical Response) Destroying call '52@192.168.0.153' Sip read: BYE sip:3218888@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153;branch=z9hG4bK2038176231 From: <sip:2000@192.168.0.203>; To: <sip:3218888@192.168.0.203> Call-ID: 52@192.168.0.153 CSeq:21 Contact= <sip:2000@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@192.168.0.203>; To: <sip:3218888@192.168.0.203>;tag=as31bcfc3e Call-ID: 52@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@192.168.0.153' __________________________________ Do you Yahoo!? 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