[SR-Users] message 484
Pedro Niño
nino.pedro at gmail.com
Wed Apr 2 02:40:58 CEST 2014
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.pedro at 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 at networklab.ca>
> escribió:
>
>> 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 at networklab.loc
>> ;transport=UDP>;tag=6358d712
>> To: <sip:120 at 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 at 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 at networklab.loc
>> ;transport=UDP>;tag=6358d712
>> To: <sip:120 at 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 at 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 at 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 at networklab.loc;transport=UDP>;tag=as3b53c4ae
>> To: "Slava Bendersky"<sip:1200 at 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 at networklab.loc;transport=UDP>;tag=6358d712
>> From: <sip:120 at 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 at networklab.loc>;tag=as7232ca20
>> To: <sip:10.237.236.207>
>> Contact: <sip:1300 at 10.237.236.207:5062>
>> Call-ID: 46ea55704ee7005705c98d9106904470 at 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 at gmail.com>
>> *To: *"Kamailio (SER) - Users Mailing List" <
>> sr-users at 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 at networklab.ca>
>> escribió:
>>
>>> 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 at edvina.net>
>>> *To: *"Kamailio (SER) - Users Mailing List" <
>>> sr-users at 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 at gmail.com> wrote:
>>>
>>> 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 at networklab.ca>
>>> escribió:
>>>
>>>> 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 at networklab.loc>;tag=as0a530a8d
>>>> To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df
>>>> ---> This line ins question.
>>>> Call-ID: 631e893f75da720865e8468132884367 at networklab.loc
>>>> CSeq: 102 OPTIONS
>>>> Contact: <sip:1300 at 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 at 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 at 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 at 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 at 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 at 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 at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
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