[SR-Users] message 484

Mon Apr 7 19:50:38 CEST 2014

Hello Pedro, 
I just come back on line. 
If i remove this line I start getting 

----- Original Message -----

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: 

<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 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: 

<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 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: 

<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 at 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 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. 

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</blockquote>

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

</blockquote>

_______________________________________________ 
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 

</blockquote>

</blockquote>

_______________________________________________ 
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