Hello Pedro, When I removing this line I starts getting "484","Address Incomplete"
I tried enable rtp debug on asterisk and look like all re transmissions cause by reinvite. Slava. ----- Original Message ----- From: "Pedro Niño" <nino.pe...@gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org> Sent: Tuesday, April 1, 2014 8:40:58 PM Subject: Re: [SR-Users] message 484 I think you should remove this section: or comment it, its behavior is not the one we want at this moment. ------- if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if (is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "OK"); } ----- El abr 1, 2014 7:58 PM, "Pedro Niño" < nino.pe...@gmail.com > escribió: Sorry, I was out for a while. Still have this issue? >From what I am seeing, asterisk is expecting for the password. Is the >voicemail configured ? Check username and password. Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" < volga...@networklab.ca > escribió: <blockquote> Hello Pedro, Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue. [voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no -- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language 'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: <sip:10.237.236.207;lr=on> From: "Slava Bendersky"<sip:1...@networklab.loc;transport=UDP>;tag=6358d712 To: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183 v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- Retransmitting #10 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: <sip:10.237.236.207;lr=on> From: "Slava Bendersky"<sip:1...@networklab.loc;transport=UDP>;tag=6358d712 To: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183 v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing <sip:10.237.236.207;lr=on> for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060 : BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: <sip:10.237.236.207;lr=on> Max-Forwards: 70 From: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae To: "Slava Bendersky"<sip:1...@networklab.loc;transport=UDP>;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0 --- <--- SIP read from UDP: 10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"<sip:1...@networklab.loc;transport=UDP>;tag=6358d712 From: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060 : OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" <sip:1...@networklab.loc>;tag=as7232ca20 To: <sip:10.237.236.207> Contact: < sip:1300@10.237.236.207:5062 > Call-ID: 46ea55704ee7005705c98d9106904...@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 Slava. From: "Pedro Niño" < nino.pe...@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 9:51:11 AM Subject: Re: [SR-Users] message 484 So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online? All the users are on the same asterisk server? Or using a trunk outside? As a test, tried to register to the asterisk server directly and test the call? That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful El mar 31, 2014 8:13 AM, "Slava Bendersky" < volga...@networklab.ca > escribió: <blockquote> Hello Olle, Overlap is disabled on asterisk. I more wonder about this message. Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri Because from direct connected network, call failing to voicemail. Slva. From: "Olle E. Johansson" < o...@edvina.net > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484 Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call". /O On 31 Mar 2014, at 02:30, Pedro Niño < nino.pe...@gmail.com > wrote: <blockquote> I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow. Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga...@networklab.ca > escribió: <blockquote> Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ? <--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1...@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884...@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </blockquote> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </blockquote> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </blockquote> </blockquote> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users