[SR-Users] message 484

Wed Apr 2 02:28:20 CEST 2014

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
>>
>>
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> sr-users at lists.sip-router.org
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>
>
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