Summary: | ASTERISK-16700: SIP Trunk ${DIALSTATUS} wrong return code - it is always "ANSWER" status | ||
Reporter: | Chernov Roman (romirikos) | Labels: | |
Date Opened: | 2010-09-20 03:30:20 | Date Closed: | 2011-06-07 14:05:06 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/NewFeature |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | 1. First I show my SIP Trunk and SIP phone user configuration 1.1 SIP Trunk configuration [WMAG006-trunk] dtmfmode=rfc2833 type=friend host=*** IP VoIP-Provider *** fromuser=*** My username *** fromdomain=*** VoIP-Provider Domain Name *** username=*** username *** secret=*** password *** insecure=port,invite canreinvite=no qualify=yes callgroup= context=WMAG006-outbounds-calls nat=yes disallow=all allow=g729 1.2 SIP phone username configuration [505] allow=all secret=505 dtmfmode=rfc2833 canreinvite=no context=WMAG006-outbounds-calls host=dynamic type=friend nat=yes port=5060 qualify=yes dial=SIP/505 permit=0.0.0.0/0.0.0.0 callerid=505 call-limit=50 faxdetect=no disallow=all allow=g729 2. Here is my DialPlan in /etc/asterisk/extensions.conf [WMAG006-outbounds-calls] ; England, Spain (Code number + phone number = 12 digits) ; exten => _XXXXXXXXXXXX,1,Set(CALLERID(all)=08000963317) ; exten => _XXXXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,25,g) exten => _XXXXXXXXXXXX,1,Dial(SIP/WMAG006-trunk/${EXTEN},5,R) exten => _XXXXXXXXXXXX,n,Hangup() ; England 13 digits ;exten => _XXXXXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,5) ;exten => _XXXXXXXXXXXXX,n,Hangup() ; USA, Australia (Code number + phone number = 11 digits) ;exten => _XXXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,5) ;exten => _XXXXXXXXXXX,n,Hangup() ; Denmark (Code number + phone number = 10 digits) ;exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,5) ;exten => _XXXXXXXXXX,n,Hangup() exten => 505,1,Answer() exten => 505,n,Dial(SIP/505,25) exten => 505,n,Hangup() exten => 501,1,Answer() exten => 501,n,Dial(SIP/501,25) exten => 501,n,Hangup() exten => h,1,Goto(s-{DIALSTATUS},1) ; exten => h,1,System(/bin/sh -c "/bin/echo ${DIALSTATUS} > /usr/local/asterisk/wmag006/status") exten => s-CANCEL,1,System(/bin/sh -c "/bin/echo Cancel > /usr/local/asterisk/wmag006/status") exten => s-ANSWER,1,System(/bin/sh -c "/bin/echo Answer > /usr/local/asterisk/wmag006/status") exten => s-NOANSWER,1,System(/bin/sh -c "/bin/echo NoAnswer > /usr/local/asterisk/wmag006/status") exten => s-BUSY,1,System(/bin/sh -c "/bin/echo Busy `date +%F-%H:%M` > /usr/local/asterisk/wmag006/status") exten => s-CONGESTION,1,System(/bin/sh -c "/bin/echo Congestion `date +%F-%H:%M` > /usr/local/asterisk/wmag006/status") exten => s-CHANUNAVAIL,1,System(/bin/sh -c "/bin/echo Chanunavail `date +%F-%H:%M` > /usr/local/asterisk/wmag006/status") 3. Debug output from Asterisk 1.8 CLI -- Registered SIP '505' at 192.168.0.7:2950 > Saved useragent "sipLite" for peer 505 [Sep 20 11:30:58] NOTICE[30357]: chan_sip.c:19477 handle_response_peerpoke: Peer '505' is now Reachable. (24ms / 2000ms) == Using SIP RTP CoS mark 5 -- Executing [380623880525@WMAG006-outbounds-calls:1] Dial("SIP/505-00000002", "SIP/WMAG006-trunk/380623880525,5,R") in new stack == Using SIP RTP CoS mark 5 -- Called WMAG006-trunk/380623880525 -- SIP/WMAG006-trunk-00000003 answered SIP/505-00000002 -- Locally bridging SIP/505-00000002 and SIP/WMAG006-trunk-00000003 -- Executing [h@WMAG006-outbounds-calls:1] System("SIP/505-00000002", "/bin/sh -c "/bin/echo ANSWER > /usr/local/asterisk/wmag006/status"") in new stack == Spawn extension (WMAG006-outbounds-calls, 380623880525, 1) exited non-zero on 'SIP/505-00000002' ------------------------------------------------------------------------- So, the issue is when I place the call through the SIP trunk it's always return - "ANSWER" status even if the other side was BUSY, NOANSWER, CONGESTION and etc. But if I place the call between (Local IP phones that connected directly to Asterisk 1.8) for example I call from 505 to 501 and it return RIGHT ${DIALSTATUS} code ****** ADDITIONAL INFORMATION ****** Please help me to fix this bug. Thanks a lot. | ||
Comments: | By: David Woolley (davidw) 2010-09-20 05:46:46 The bug reporting guidelines require that you provide sip set debug output. I suspect that will show that the downstream system has answered the call before sending call progress indications, in which case you have a support issue, not a bug. .../NewFeature is not the right category, and your descriptions suggests that you have tried and it is repeatable. By: Chernov Roman (romirikos) 2010-09-21 04:48:17 This is Full SIP Debug output SIP Debugging enabled Reliably Transmitting (NAT) to 192.168.2.27:5060: OPTIONS sip:503@192.168.2.27:5060;line=3aehmciw SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK6b21a27b;rport From: "asterisk" <sip:asterisk@192.168.2.1>;tag=as1ec1276f To: <sip:503@192.168.2.27:5060;line=3aehmciw> Contact: <sip:asterisk@192.168.2.1> Call-ID: 589b67ab6b35dd0f33901fd56ea54d54@192.168.2.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Sep 2010 09:39:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.2.27:5060 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK6b21a27b;rport=5060 f: "asterisk" <sip:asterisk@192.168.2.1>;tag=as1ec1276f t: <sip:503@192.168.2.27:5060;line=3aehmciw> i: 589b67ab6b35dd0f33901fd56ea54d54@192.168.2.1 CSeq: 102 OPTIONS m: <sip:27@192.168.2.27:5060;line=ydg4eu9f>;flow-id=1 User-Agent: snom320/6.5.4 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid l: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '589b67ab6b35dd0f33901fd56ea54d54@192.168.2.1' Method: OPTIONS <--- SIP read from 192.168.2.23:5060 ---> INVITE sip:380623880525@195.184.196.222 SIP/2.0 v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9fqz7t2ezr3j;rport f: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr t: <sip:380623880525@195.184.196.222> i: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 1 INVITE Max-Forwards: 30 m: <sip:505@192.168.2.23:5060;line=x0v8xene>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 c: application/sdp l: 475 v=0 o=root 1120826698 1120826698 IN IP4 192.168.2.23 s=call c=IN IP4 192.168.2.23 t=0 0 m=audio 22426 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:btyoi9sVrX6ST6gnWRVRMmff/STy+Xbvc+in/h+s a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 19 lines) --- Sending to 192.168.2.23 : 5060 (no NAT) Using INVITE request as basis request - 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 <--- Reliably Transmitting (NAT) to 192.168.2.23:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9fqz7t2ezr3j;received=192.168.2.23;rport=5060 From: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr To: <sip:380623880525@195.184.196.222>;tag=as69eb4b51 Call-ID: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d4ab296" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c267b928b29-b7et3f97hqmi@snom320-0004132485B0' in 32000 ms (Method: INVITE) Found user '505' <--- SIP read from 192.168.2.23:5060 ---> ACK sip:380623880525@195.184.196.222 SIP/2.0 v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9fqz7t2ezr3j;rport f: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr t: <sip:380623880525@195.184.196.222>;tag=as69eb4b51 i: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 1 ACK Max-Forwards: 30 m: <sip:505@192.168.2.23:5060;line=x0v8xene>;flow-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 192.168.2.23:5060 ---> INVITE sip:380623880525@195.184.196.222 SIP/2.0 v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9ov4p7niuvbq;rport f: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr t: <sip:380623880525@195.184.196.222> i: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 2 INVITE Max-Forwards: 30 m: <sip:505@192.168.2.23:5060;line=x0v8xene>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="505",realm="asterisk",nonce="1d4ab296",uri="sip:380623880525@195.184.196.222",response="28ec0950e18d05f656a39e57d30cdd11",algorithm=md5 c: application/sdp l: 475 v=0 o=root 1120826698 1120826698 IN IP4 192.168.2.23 s=call c=IN IP4 192.168.2.23 t=0 0 m=audio 22426 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:btyoi9sVrX6ST6gnWRVRMmff/STy+Xbvc+in/h+s a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 19 lines) --- Sending to 192.168.2.23 : 5060 (NAT) Using INVITE request as basis request - 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 Found user '505' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format g722 for ID 9 Found audio description format g726-32 for ID 2 Found audio description format gsm for ID 3 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.2.23:22426 Looking for 380623880525 in WMAG006-outbounds-calls (domain 195.184.196.222) list_route: hop: <sip:505@192.168.2.23:5060;line=x0v8xene> <--- Transmitting (NAT) to 192.168.2.23:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9ov4p7niuvbq;received=192.168.2.23;rport=5060 From: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr To: <sip:380623880525@195.184.196.222> Call-ID: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:380623880525@192.168.2.1> Content-Length: 0 <------------> -- Executing [380623880525@WMAG006-outbounds-calls:1] Dial("SIP/505-00000008", "SIP/380623880525@WMAG006-trunk") in new stack Audio is at 195.184.196.222 port 16174 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 213.144.184.12:5060: INVITE sip:380623880525@foib.future-b.eu SIP/2.0 Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 To: <sip:380623880525@foib.future-b.eu> Contact: <sip:WMAG006@195.184.196.222> Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu CSeq: 102 INVITE ser-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Sep 2010 09:40:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 269 v=0 o=root 17187 17187 IN IP4 195.184.196.222 s=session c=IN IP4 195.184.196.222 t=0 0 m=audio 16174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 380623880525@WMAG006-trunk <--- SIP read from 213.144.184.12:5060 ---> SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu To: <sip:380623880525@foib.future-b.eu> Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 213.144.184.12:5060 ---> SIP/2.0 407 Proxy Authentication Required CSeq: 102 INVITE Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu To: <sip:380623880525@foib.future-b.eu>;tag=1037221576942191 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="15769421917680812821101026409372238238" Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 213.144.184.12:5060: ACK sip:380623880525@foib.future-b.eu SIP/2.0 Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 To: <sip:380623880525@foib.future-b.eu>;tag=1037221576942191 Contact: <sip:WMAG006@195.184.196.222> Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 195.184.196.222 port 16174 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 213.144.184.12:5060: INVITE sip:380623880525@foib.future-b.eu SIP/2.0 Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK07fef3d8;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 To: <sip:380623880525@foib.future-b.eu> Contact: <sip:WMAG006@195.184.196.222> Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="WMAG006", realm="VoipSwitch", algorithm=MD5, uri="sip:380623880525@foib.future-b.eu", nonce="15769421917680812821101026409372238238", response="bd0278f6c9f6cb162d92abefa36ac70f" Date: Tue, 21 Sep 2010 09:40:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 269 v=0 o=root 17187 17188 IN IP4 195.184.196.222 s=session c=IN IP4 195.184.196.222 t=0 0 m=audio 16174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 213.144.184.12:5060 ---> SIP/2.0 100 Trying CSeq: 103 INVITE Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK07fef3d8;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu To: <sip:380623880525@foib.future-b.eu> Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 213.144.184.12:5060 ---> SIP/2.0 200 OK CSeq: 103 INVITE Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK07fef3d8;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu To: <sip:380623880525@foib.future-b.eu>;tag=1037221576942347 Contact: <sip:213.144.184.12:5060;transport=udp> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER Content-Type: application/sdp Content-Length: 236 v=0 o=- 1654832215 1576942191 IN IP4 213.144.184.12 s=VoipSIP c=IN IP4 213.144.184.12 t=0 0 m=audio 6032 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (10 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.144.184.12:6032 list_route: hop: <sip:213.144.184.12:5060;transport=udp> set_destination: Parsing <sip:213.144.184.12:5060;transport=udp> for address/port to send to set_destination: set destination to 213.144.184.12, port 5060 Transmitting (NAT) to 213.144.184.12:5060: ACK sip:213.144.184.12:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK43d609a2;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 To: <sip:380623880525@foib.future-b.eu>;tag=1037221576942347 Contact: <sip:WMAG006@195.184.196.222> Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/WMAG006-trunk-00000009 answered SIP/505-00000008 Audio is at 192.168.2.1 port 17028 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.2.23:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9ov4p7niuvbq;received=192.168.2.23;rport=5060 From: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr To: <sip:380623880525@195.184.196.222>;tag=as0db2ebea Call-ID: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:380623880525@192.168.2.1> Content-Type: application/sdp Content-Length: 261 v=0 o=root 17187 17187 IN IP4 192.168.2.1 s=session c=IN IP4 192.168.2.1 t=0 0 m=audio 17028 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/505-00000008 and SIP/WMAG006-trunk-00000009 <--- SIP read from 192.168.2.23:5060 ---> ACK sip:380623880525@192.168.2.1 SIP/2.0 v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-10bi0abz0u6e;rport f: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr t: <sip:380623880525@195.184.196.222>;tag=as0db2ebea i: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 2 ACK Max-Forwards: 30 m: <sip:505@192.168.2.23:5060;line=x0v8xene>;flow-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- Reliably Transmitting (NAT) to 192.168.2.11:5060: OPTIONS sip:504@192.168.2.11:5060;line=ty8vwu7u SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK31b18e1c;rport From: "asterisk" <sip:asterisk@192.168.2.1>;tag=as42b34502 To: <sip:504@192.168.2.11:5060;line=ty8vwu7u> Contact: <sip:asterisk@192.168.2.1> Call-ID: 36e3b8fe6974a432044047a76a834d6b@192.168.2.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Sep 2010 09:40:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.2.11:5060 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK31b18e1c;rport=5060 f: "asterisk" <sip:asterisk@192.168.2.1>;tag=as42b34502 t: <sip:504@192.168.2.11:5060;line=ty8vwu7u> i: 36e3b8fe6974a432044047a76a834d6b@192.168.2.1 CSeq: 102 OPTIONS m: <sip:11@192.168.2.11:5060;line=lhdez2b1>;flow-id=1 User-Agent: snom320/6.2.2 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid l: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '36e3b8fe6974a432044047a76a834d6b@192.168.2.1' Method: OPTIONS <--- SIP read from 192.168.2.23:5060 ---> BYE sip:380623880525@192.168.2.1 SIP/2.0 v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-wld78hghc563;rport f: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr t: <sip:380623880525@195.184.196.222>;tag=as0db2ebea i: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 3 BYE Max-Forwards: 30 m: <sip:505@192.168.2.23:5060;line=x0v8xene>;flow-id=1 User-Agent: snom320/6.2.2 RTP-RxStat: Total_Rx_Pkts=592,Rx_Pkts=592,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=722,Tx_Pkts=722,Remote_Tx_Pkts=0 l: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.2.23 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.2.23:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-wld78hghc563;received=192.168.2.23;rport=5060 From: "505" <sip:505@195.184.196.222>;tag=0oc4ee71xr To: <sip:380623880525@195.184.196.222>;tag=as0db2ebea Call-ID: 3c267b928b29-b7et3f97hqmi@snom320-0004132485B0 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> -- Executing [h@WMAG006-outbounds-calls:1] System("SIP/505-00000008", "/bin/sh -c "/bin/echo ANSWER > /usr/local/asterisk/wmag006/status"") in new stack Scheduling destruction of SIP dialog '356e350427ca0d40294efae962c045f5@foib.future-b.eu' in 9152 ms (Method: INVITE) set_destination: Parsing <sip:213.144.184.12:5060;transport=udp> for address/port to send to set_destination: set destination to 213.144.184.12, port 5060 Reliably Transmitting (NAT) to 213.144.184.12:5060: BYE sip:213.144.184.12:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK325599a1;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 To: <sip:380623880525@foib.future-b.eu>;tag=1037221576942347 Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="WMAG006", realm="VoipSwitch", algorithm=MD5, uri="sip:213.144.184.12:5060", nonce="15769421917680812821101026409372238238", response="adf62a8a2498997b44166566a3e70f5b" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (WMAG006-outbounds-calls, 380623880525, 1) exited non-zero on 'SIP/505-00000008' <--- SIP read from 213.144.184.12:5060 ---> SIP/2.0 200 OK CSeq: 104 BYE Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK325599a1;rport From: "505" <sip:WMAG006@foib.future-b.eu>;tag=as51bae2b2 Call-ID: 356e350427ca0d40294efae962c045f5@foib.future-b.eu To: <sip:380623880525@foib.future-b.eu>;tag=1037221576942347 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '356e350427ca0d40294efae962c045f5@foib.future-b.eu' Method: INVITE Really destroying SIP dialog '3c267b928b29-b7et3f97hqmi@snom320-0004132485B0' Method: BYE tele2web-ukraine*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk ending (0). By: David Woolley (davidw) 2010-09-21 04:56:34 The remote system has said that the call is answered. This is not a bug.in Asterisk. <--- SIP read from 213.144.184.12:5060 ---> SIP/2.0 200 OK CSeq: 103 INVITE You should first try to get the remote system to fix their system. If that fails, you need to use a support (mailing list, forum, or IRC) channel for asterisk, not a bug reporting one, to investigate possible workarounds. By: Paul Belanger (pabelanger) 2010-09-21 13:52:04 Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker, http://www.asterisk.org/support |