[SR-Users] message 484
Pedro Niño
nino.pedro at gmail.com
Wed Apr 9 20:35:01 CEST 2014
Getting....?
El abr 7, 2014 1:21 PM, "Slava Bendersky" <volga629 at networklab.ca>
escribió:
> Hello Pedro,
> I just come back on line.
> If i remove this line I start getting
>
>
> ------------------------------
> *From: *"Pedro Niño" <nino.pedro at gmail.com>
> *To: *"Kamailio (SER) - Users Mailing List" <sr-users at 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.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
>>>
>>>
> _______________________________________________
> 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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