Hello Pedro, I just come back on line. If i remove this line I start getting
----- Original Message -----
From: "Pedro Niño" nino.pedro@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.
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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.pedro@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" < volga629@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:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@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:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@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:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@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:1200@networklab.loc;transport=UDP;tag=6358d712 From: sip:120@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:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: < sip:1300@10.237.236.207:5062 > Call-ID: 46ea55704ee7005705c98d9106904470@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.pedro@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" < volga629@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" < oej@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.pedro@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" < volga629@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:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@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.
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_______________________________________________ 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>
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</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>
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