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 sip%3A119@192.168.10.1>;tag=7dc2f363e7b65552 To: <sip:114@192.168.10.1 sip%3A114@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 sip%3A119@192.168.10.1>;tag=7dc2f363e7b65552 Call-ID: e3b32187ee145e96@192.168.10.28 To: <sip:114@192.168.10.1 sip%3A114@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@192.168.10.1>;tag=d096ebe220002321 To: <sip:111@192.168.10.1 sip%3A111@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%3A111@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@192.168.10.1>;tag=45d17c11e5a1f791 To: <sip:111@192.168.10.1 sip%3A111@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 sip%3A119@192.168.10.1>;tag=45d17c11e5a1f791 To: <sip:111@192.168.10.1 sip%3A111@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%3A111@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@192.168.10.1>;tag=45d17c11e5a1f791 To: <sip:111@192.168.10.1 sip%3A111@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%3A111@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@192.168.10.1>;tag=45d17c11e5a1f791 To: <sip:111@192.168.10.1 sip%3A111@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@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
Como has hecho la vista de la tabla subscriber? Yo tuve más de un problema con eso...
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@gmail.com
Como has hecho la vista de la tabla subscriber? Yo tuve más de un problema con eso...
En principio parece estar OK. Como has configurado las opciones de cacheo de RealTime?
2008/10/14 Saúl Ibarra saghul@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
2008/10/14 troxlinux xserverlinux@gmail.com:
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.
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
sr-users-es@lists.kamailio.org
-
Iñaki Baz Castillo -
Saúl Ibarra -
troxlinux