[SR-Users] message 484

Slava Bendersky volga629 at networklab.ca
Mon Mar 31 16:09:54 CEST 2014


Hello Pedro, 
Asterisk is used only for voicemail. All extension are terminates and registrar on kamailio gateway. Kamailio used ldap and registration working ok. I can come back to lab soon after meeting and I post some debug. 


Slava. 

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

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: 


<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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_______________________________________________ 
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</blockquote>



_______________________________________________ 
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_______________________________________________ 
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</blockquote>


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