telefonos 3COM 3101SP y migracion de asterisk

[IOCAPP]

Hola buen dia Lista como estan?

tengo un telefono 3com 3101SP que se le carga el firmware de SIP sin ningun problema y este se conecta a un asterisk version 1.4

pero al actuaizar la version del asterisk ya sea a una version 1.8 10 11 o 12 siempre se comporta de la siguiente manera

me manda este mensaje

CSeq: 112 REGISTER

esta es una traza

[Jan 30 10:54:18] VERBOSE[1573] chan_sip.c: --- (11 headers 0 lines) ---
[Jan 30 10:54:19] VERBOSE[1573] chan_sip.c:
�<--- SIP read from UDP:10.0.200.205:5060 --->
�REGISTER sip:10.0.200.203 SIP/2.0
�v: SIP/2.0/UDP 10.0.200.205:5060
�t: <sip:00e0bbb347df@10.0.200.203>
�f: <sip:00e0bbb347df@10.0.200.203>
�i: fffe1438-01d6-1fd2-13c5-00e0bbb347df
�CSeq: 110 REGISTER
�Max-Forwards: 70
�m: <sip:00e0bbb347df@10.0.200.205:5060>;dt=546;sn=TLC 98N7B347DF;mac=00e0bbb347df;ver=X9.5.13.13
�Expires: 0
�User-Agent: 3Com-SIP-Phone/V9.5.13.13;mac=00e0bbb347df
�Event: x-device-reg;dt=546
�
�<------------->
[Jan 30 10:54:19] VERBOSE[1573] chan_sip.c: --- (11 headers 0 lines) ---
[Jan 30 10:54:21] VERBOSE[1573] chan_sip.c:
�<--- SIP read from UDP:10.0.0.150:5060 --->
�OPTIONS sip:10.0.200.203 SIP/2.0
�Via: SIP/2.0/UDP 10.0.0.150:5060;branch=z9hG4bK57b9d0e1
�Max-Forwards: 70
�From: "asterisk" <sip:asterisk@10.0.0.150>;tag=as49043f83
�To: <sip:10.0.200.203>
�Contact: <sip:asterisk@10.0.0.150:5060>
�Call-ID: 435820e10dbba65840f693d822b047db@10.0.0.150:5060
�CSeq: 102 OPTIONS
�User-Agent: Asterisk PBX 1.8.17.0
�Date: Thu, 30 Jan 2014 16:56:21 GMT
�Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
�Supported: replaces, timer
�Content-Length: 0
�
�<------------->
[Jan 30 10:54:21] VERBOSE[1573] chan_sip.c: --- (13 headers 0 lines) ---
[Jan 30 10:54:21] VERBOSE[1573] chan_sip.c: Sending to 10.0.0.150:5060 (NAT)
[Jan 30 10:54:21] VERBOSE[1573] chan_sip.c: Looking for s in from-sip (domain 10.0.200.203)
[Jan 30 10:54:21] VERBOSE[1573] chan_sip.c:
�<--- Transmitting (NAT) to 10.0.0.150:5060 --->
�SIP/2.0 404 Not Found
�Via: SIP/2.0/UDP 10.0.0.150:5060;branch=z9hG4bK57b9d0e1;received=10 .0.0.150;rport=5060
�From: "asterisk" <sip:asterisk@10.0.0.150>;tag=as49043f83
�To: <sip:10.0.200.203>;tag=as0f1c5df4
�Call-ID: 435820e10dbba65840f693d822b047db@10.0.0.150:5060
�CSeq: 102 OPTIONS
�Server: Asterisk PBX 11.7.0
�Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
�Supported: replaces, timer
�Accept: application/sdp
�Content-Length: 0
�
�
�<------------>
[Jan 30 10:54:21] VERBOSE[1573] chan_sip.c: Scheduling destruction of SIP dialog '435820e10dbba65840f693d822b047db@10.0.0.150:5060' in 32000 ms (Method: OPTIONS)
[Jan 30 10:54:21] VERBOSE[1573] chan_sip.c:
�<--- SIP read from UDP:10.0.200.205:5060 --->
�REGISTER sip:10.0.200.203 SIP/2.0
�v: SIP/2.0/UDP 10.0.200.205:5060
�t: <sip:00e0bbb347df@10.0.200.203>
�f: <sip:00e0bbb347df@10.0.200.203>
�i: fffe1438-01d6-1fd2-13c5-00e0bbb347df
�CSeq: 110 REGISTER
�Max-Forwards: 70
�m: <sip:00e0bbb347df@10.0.200.205:5060>;dt=546;sn=TLC 98N7B347DF;mac=00e0bbb347df;ver=X9.5.13.13
�Expires: 0
�User-Agent: 3Com-SIP-Phone/V9.5.13.13;mac=00e0bbb347df
�Event: x-device-reg;dt=546
�m: <sip:00e0bbb347df@10.0.200.205:5060>;dt=546;sn=TLC 98N7B347DF;mac=00e0bbb347df;ver=X9.5.13.13
�Expires: 0
�User-Agent: 3Com-SIP-Phone/V9.5.13.13;mac=00e0bbb347df
�Event: x-device-reg;dt=546
�




sin embargo este mismo telefono con asterisk 1.4 se conecta sin problema---

[Jan 30 05:33:03] VERBOSE[1370] logger.c:
<--- SIP read from 10.0.0.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.200.203:5060;branch=z9hG4bK54157d4e;received= 10.0.200.203;rport=5060
From: "asterisk" <sip:asterisk@10.0.200.203>;tag=as732ea364
To: <sip:10.0.0.150>;tag=as09b491ca
Call-ID: 4223eb9d3107287b5f75f0741909b9f3@10.0.200.203
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:10.0.0.150:5060>
Accept: application/sdp
Content-Length: 0


<------------->
[Jan 30 05:33:03] VERBOSE[1370] logger.c: --- (12 headers 0 lines) ---
[Jan 30 05:33:03] VERBOSE[1370] logger.c: Really destroying SIP dialog '4223eb9d3107287b5f75f0741909b9f3@10.0.200.203' Method: OPTIONS
[Jan 30 05:33:05] VERBOSE[1370] logger.c:
<--- SIP read from 10.0.0.150:5060 --->
OPTIONS sip:10.0.200.203 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.150:5060;branch=z9hG4bK1e22641a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.150>;tag=as6c484323
To: <sip:10.0.200.203>
Contact: <sip:asterisk@10.0.0.150:5060>
Call-ID: 0934e96f73ab71b53b03370870766c98@10.0.0.150:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.17.0
Date: Thu, 30 Jan 2014 15:35:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------->
[Jan 30 05:33:05] VERBOSE[1370] logger.c: --- (13 headers 0 lines) ---
[Jan 30 05:33:05] VERBOSE[1370] logger.c: Looking for s in from-sip (domain 10.0.200.203)
[Jan 30 05:33:05] VERBOSE[1370] logger.c:
<--- Transmitting (no NAT) to 10.0.0.150:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.150:5060;branch=z9hG4bK1e22641a;received=10 .0.0.150
From: "asterisk" <sip:asterisk@10.0.0.150>;tag=as6c484323
To: <sip:10.0.200.203>;tag=as625adfe5
Call-ID: 0934e96f73ab71b53b03370870766c98@10.0.0.150:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[Jan 30 05:33:05] VERBOSE[1370] logger.c: Scheduling destruction of SIP dialog '0934e96f73ab71b53b03370870766c98@10.0.0.150:5060' in 3200
0 ms (Method: OPTIONS)
[Jan 30 05:33:15] VERBOSE[1370] logger.c: Reliably Transmitting (no NAT) to 10.0.200.205:5060:
OPTIONS sip:5000@10.0.200.205:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.200.203:5060;branch=z9hG4bK6ebbc87b;rport
From: "asterisk" <sip:asterisk@10.0.200.203>;tag=as66fd1528
To: <sip:5000@10.0.200.205:5060>
Contact: <sip:asterisk@10.0.200.203>
Call-ID: 23bf347219a1dada62c681df523c3e8d@10.0.200.203
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 30 Jan 2014 09:33:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Jan 30 05:33:15] VERBOSE[1370] logger.c:
<--- SIP read from 10.0.200.205:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 10.0.200.203:5060;branch=z9hG4bK6ebbc87b;rport
t: <sip:5000@10.0.200.205:5060>;tag=967f1728-01d6-1fa2-13cb-00e0bbb347df
f: "asterisk" <sip:asterisk@10.0.200.203>;tag=as66fd1528
i: 23bf347219a1dada62c681df523c3e8d@10.0.200.203
CSeq: 102 OPTIONS
m: <sip:5000@10.0.200.205:5060>
Allow: INVITE,BYE,CANCEL,OPTIONS,NOTIFY,REFER
P-Asserted-Identity: "" <sip:5000@10.0.200.203>
l: 0



este es mi sip.conf en ambos casos

[general]
port = 5060 ;
bindport = 5060
bindaddr=10.0.200.203 ;
context=from-sip ;
;allowguest=no
;alwaysauthreject=yes
dtmfmode=rfc2833 ; Tonos DTMF
allow=all ;
allow=h263
allow=h263p
videosupport=yes
t38pt_udptl=yes
localnet=10.0.0.0/255.255.0.0

[5000]
type=friend
;type=peer
username=5000
callerid="SIP Phone" <5000>
secret=
host=dynamic
defaultip=10.0.200.205
deny=0.0.0.0/0.0.0.0
permit=10.0.200.205/255.255.0.0
record_out=Adhoc
record_in=Adhoc
qualify=yes
context=from-sip
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=g729
nat=no
dtmfmode=rfc2833
mailbox=5000@default
canreinvite=no
reinvite=no
transfer=yes
callgroup=1
pickupgroup=1


Espero me puedan ayudar con sus aportaciones para resolver este problema Saludos

[hector]

ponle un password (secret) a esa extensión ... y en el trazado del 1.4 te falta el primer dialogo del SIP (no viene el INVITE) para poder revisarlo mejor