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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153;branch=z9hG4bK2038176231 From:<sip:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.153;branch=z9hG4bK2038176231 From: <sip:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32> Call-ID: [EMAIL PROTECTED] CSeq: 22 INFO Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 22 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32> Call-ID: [EMAIL PROTECTED] CSeq: 23 INFO Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 23 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32> Call-ID: [EMAIL PROTECTED] CSeq: 24 INFO Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 24 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32> Call-ID: [EMAIL PROTECTED] CSeq: 25 INFO Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]> To: <sip:172.16.0.32>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 25 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:[EMAIL PROTECTED]> To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:[EMAIL PROTECTED]> To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:[EMAIL PROTECTED]> To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:[EMAIL PROTECTED]> To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:[EMAIL PROTECTED]> To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as61b12c41 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153 From: <sip:[EMAIL PROTECTED]> To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 21 (Non-critical Response) Destroying call '[EMAIL PROTECTED]' Sip read: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153;branch=z9hG4bK2038176231 From: <sip:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq:21 Contact= <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>; To: <sip:[EMAIL PROTECTED]>;tag=as31bcfc3e Call-ID: [EMAIL PROTECTED] 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 '[EMAIL PROTECTED]' __________________________________ Do you Yahoo!? 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