señores estoy integrando mi openser con asterisk realtime pues ya realice todos los pasos de asterisk para el realtime, la idea es conseguir la lucecita del voicemail , veo dentro de la base de datos de asterisk la replicacion de los usuarios de la tabla subscriber de openser todo bien aqui , lo que si me parece extraño ahora es que todas la funcionalidades que tengo con el asterisk como media server no me trabajan al 100% como las tenia antes ,por ejemplo antes cuando no contestaba me saltaba al voicemail y al integrar el realtime con asterisk me tira este error :
chan_sip.c:5095 process_sdp: Unable to lookup host in c= line, 'IN IP4 192.168.1.64192.168.1.64'
lo extraño es que esa ip en log de asterisk es la ip que vieve del dsl y no por la interfaz lan que es donde registro mis usuarios ..
algo que me este faltando del realtime? ..
saludos lista
rickygm
### sip debug asterisk ####
--- (18 headers 24 lines) ---
Sending to 192.168.10.1 : 5060 (NAT)
Using INVITE request as basis request - e3b32187ee145e96@192.168.10.28
Found user '119'
[Oct 13 23:30:23] WARNING[9912]: chan_sip.c:5095 process_sdp: Unable to lookup host in c= line, 'IN IP4 192.168.1.64192.168.1.64'
<--- Reliably Transmitting (NAT) to 192.168.10.1:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK91af.827e4ad4.1;received=192.168.10.1
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK1a8140c1b66271b0
From: <sip:119@192.168.10.1>;tag=7dc2f363e7b65552
To: <sip:114@192.168.10.1>;tag=as5a93f32a
Call-ID: e3b32187ee145e96@192.168.10.28
CSeq: 25247 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'e3b32187ee145e96@192.168.10.28' in 32000 ms (Method: INVITE)
xserver*CLI>
<--- SIP read from 192.168.10.1:5060 --->
ACK sip:u114@192.168.10.1:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK91af.827e4ad4.1
From: <sip:119@192.168.10.1>;tag=7dc2f363e7b65552
Call-ID: e3b32187ee145e96@192.168.10.28
To: <sip:114@192.168.10.1>;tag=as5a93f32a
CSeq: 25247 ACK
Max-Forwards: 70
User-Agent: OpenSER
### mi log sip Openser ####
U +1.244159 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKb7a454e6f1909ec6
From: <sip:119@192.168.10.1>;tag=d096ebe220002321
To: <sip:111@192.168.10.1>;tag=as0cd8584b
Call-ID: 3bb16f05ffa03a24@192.168.10.28
CSeq: 30320 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
#
U +0.172349 192.168.10.28:5060 -> 192.168.10.1:5060
INVITE sip:111@192.168.10.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKf306872728753147
From: <sip:119@192.168.10.1>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1>
Contact: <sip:119@192.168.10.28:5060>
Supported: replaces, timer, path
Call-ID: 3bd1333498b35042@192.168.10.28
CSeq: 26761 INVITE
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 545
v=0
o=119 8000 8000 IN IP4 192.168.10.28
s=SIP Call
c=IN IP4 192.168.10.28
t=0 0
m=audio 5004 RTP/AVP 0 18 4 3 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 5006 RTP/AVP 99 34
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4BryA=
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2 MaxBR=1280
a=framerate:20
#
U +0.000232 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKf306872728753147;rport=5060
From: <sip:119@192.168.10.1>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4555
Call-ID: 3bd1333498b35042@192.168.10.28
CSeq: 26761 INVITE
Proxy-Authenticate: Digest realm="192.168.10.1", nonce="48f427bd23837102189868c302984a0314f3ac5e"
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
#
U +0.001676 192.168.10.28:5060 -> 192.168.10.1:5060
ACK sip:111@192.168.10.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKf306872728753147
From: <sip:119@192.168.10.1>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4555
Contact: <sip:119@192.168.10.28:5060>
Call-ID: 3bd1333498b35042@192.168.10.28
CSeq: 26761 ACK
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.001321 192.168.10.28:5060 -> 192.168.10.1:5060
INVITE sip:111@192.168.10.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK5bb5e1738e335481
From: <sip:119@192.168.10.1>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1>
Contact: <sip:119@192.168.10.28:5060>
Supported: replaces, timer, path
Proxy-Authorization: Digest username="119", realm="192.168.10.1", algorithm=MD5, uri="sip:111@192.168.10.1", nonce="48f427bd23837102189868c302984a0314f3ac5e", response="d5d20ad6310dea0460b3f4ca8b020e27"
Call-ID: 3bd1333498b35042@192.168.10.28
CSeq: 26762 INVITE
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 545
v=0
o=119 8000 8001 IN IP4 192.168.10.28
s=SIP Call
c=IN IP4 192.168.10.28
t=0 0
m=audio 5004 RTP/AVP 0 18 4 3 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 5006 RTP/AVP 99 34
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4BryA=
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2 MaxBR=1280
a=framerate:20