14 Oct
2008
14 Oct
'08
9:32 a.m.
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(a)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 <sip%3A119(a)192.168.10.1>>;tag=7dc2f363e7b65552
To: <sip:114@192.168.10.1 <sip%3A114(a)192.168.10.1>>;tag=as5a93f32a
Call-ID: e3b32187ee145e96(a)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(a)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 <sip%3A119(a)192.168.10.1>>;tag=7dc2f363e7b65552
Call-ID: e3b32187ee145e96(a)192.168.10.28
To: <sip:114@192.168.10.1 <sip%3A114(a)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 <sip%3A119(a)192.168.10.1>>;tag=d096ebe220002321
To: <sip:111@192.168.10.1 <sip%3A111(a)192.168.10.1>>;tag=as0cd8584b
Call-ID: 3bb16f05ffa03a24(a)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%3A111(a)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 <sip%3A119(a)192.168.10.1>>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1 <sip%3A111(a)192.168.10.1>>
Contact: <sip:119@192.168.10.28:5060>
Supported: replaces, timer, path
Call-ID: 3bd1333498b35042(a)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 <sip%3A119(a)192.168.10.1>>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1
<sip%3A111(a)192.168.10.1>>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4555
Call-ID: 3bd1333498b35042(a)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%3A111(a)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 <sip%3A119(a)192.168.10.1>>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1
<sip%3A111(a)192.168.10.1>>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4555
Contact: <sip:119@192.168.10.28:5060>
Call-ID: 3bd1333498b35042(a)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%3A111(a)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 <sip%3A119(a)192.168.10.1>>;tag=45d17c11e5a1f791
To: <sip:111@192.168.10.1 <sip%3A111(a)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 <sip%3A111(a)192.168.10.1>"1>",
nonce="48f427bd23837102189868c302984a0314f3ac5e",
response="d5d20ad6310dea0460b3f4ca8b020e27"
Call-ID: 3bd1333498b35042(a)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
14 Oct
14 Oct
10:43 a.m.
New subject: [Kamailio-Users-ES] dudas integración realtime con openser
Como has hecho la vista de la tabla subscriber? Yo tuve más de un
problema con eso...
--
Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes."
----------------------------------------------------------------
http://www.saghul.net/
11:07 a.m.
New subject: [Kamailio-Users-ES] dudas integración realtime con openser
lo que hice realmente fue un ALTER TABLE subscriber ADD vmail_password
varchar(32); dentro de la base de datos de openser , lo raro es que al hacer
un sip show peers en asterisk solo me presenta el usuario de openser que
tengo creado dentro del sip.conf , y deberian aparecerme todos los usuarios
de openser , es asi como he estado leyendo ..
ojo te muestro como cree las tablas del realtime en mysql
create database asterisk;
use asterisk;
CREATE TABLE `voicemessages` (
`id` int(11) NOT NULL auto_increment,
`msgnum` int(11) NOT NULL default '0',
`dir` varchar(80) default '',
`context` varchar(80) default '',
`macrocontext` varchar(80) default '',
`callerid` varchar(40) default '',
`origtime` varchar(40) default '',
`duration` varchar(20) default '',
`mailboxuser` varchar(80) default '',
`mailboxcontext` varchar(80) default '',
`recording` longblob,
PRIMARY KEY (`id`),
KEY `dir` (`dir`)
) ENGINE=InnoDB;
CREATE VIEW vmusers AS
SELECT id as uniqueid,
username as customer_id,
'netsoluciones' as context,
username as mailbox,
vmail_password as password,
CONCAT(first_name,' ',last_name) as fullname,
email_address as email,
NULL as pager,
datetime_created as stamp
FROM openser.subscriber;
CREATE VIEW sipusers AS
SELECT username as name,
username,
'friend' as type,
NULL as secret,
domain as host,
CONCAT(rpid, ' ','<',username,'>') as callerid,
'netsoluciones' as context,
username as mailbox,
'yes' as nat,
'no' as qualify,
username as fromuser,
NULL as authuser,
domain as fromdomain,
NULL as insecure,
'no' as canreinvite,
NULL as disallow,
NULL as allow,
NULL as restrictcid,
domain as defaultip,
domain as ipaddr,
'5060' as port,
NULL as regseconds
FROM openser.subscriber;
saludoss
rickygm
2008/10/14 Saúl Ibarra <saghul(a)gmail.com>
Como has hecho la vista de la tabla subscriber? Yo
tuve más de un
problema con eso...
11:23 a.m.
New subject: [Kamailio-Users-ES] dudas integración realtime con openser
En principio parece estar OK. Como has configurado las opciones de
cacheo de RealTime?
--
Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes."
----------------------------------------------------------------
http://www.saghul.net/
7:39 p.m.
New subject: [Kamailio-Users-ES] dudas integración realtime con openser
2008/10/14 Saúl Ibarra <saghul(a)gmail.com>
En principio parece estar OK. Como has configurado las
opciones de
cacheo de RealTime?
bueno dentro del sip.conf , he agregado este par de opciones
rtcachefriends=yes
ignoreregexpire=yes
que ignoro o me falta algo mas ?
saludos estimados ..
rickygm
12:24 p.m.
New subject: [Kamailio-Users-ES] dudas integración realtime con openser
2008/10/14 troxlinux <xserverlinux(a)gmail.com>om>:
chan_sip.c:5095 process_sdp: Unable to lookup host in
c= line, 'IN IP4
192.168.1.64192.168.1.64'
¿Estaś usando RtpProxy o MediaProxy? ese parece el típico error de
sobreescribir 2 veces el SDP durante el procesado del request en
Kamailio.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
7:37 p.m.
New subject: [Kamailio-Users-ES] dudas integración realtime con openser
2008/10/14 Iñaki Baz Castillo escribio:
¿Estaś usando RtpProxy o MediaProxy?
tengo rtpproxy con openser ..
ese parece el típico error de
sobreescribir 2 veces el SDP durante el procesado del request en
Kamailio.
aqui si estoy un poco confuso, te refieres cuando trato el onreply_route ..
saludoss
rickygm
5871
days inactive
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days old
sr-users-es@lists.kamailio.org
6 comments
3 participants
participants (3)
-
Iñaki Baz Castillo -
Saúl Ibarra -
troxlinux