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:

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:

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:

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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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

_______________________________________________
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sr-users@lists.sip-router.org
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sr-users@lists.sip-router.org
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