[Kamailio-Users-ES] [OpenSER-Users-ES] Puertos asterisk y openser

[troxlinux]

2008/8/21 Iñaki Baz Castillo <ibc at aliax.net>:

> Hola, he intentado seguir el hilo pero está un poco borroso. ¿Podrías
> por favor explicar de nuevo lo que ocurre?
>

ok , Iñaki tengo openser 1.3.2 en la misma pc con asterisk - openser
escuchando en el puerto 5060 y asterisk en el 5070 , la idea es hacer
llamadas a la pstn , osea hacer de asterisk como media server y
gateway pstn, con un poco de ayuda que he recibido de la lista estoy
como que quiere , veo que cuando hago una llamada a la pstn de 7
dígitos el asterisk la recibe y la envía , lo veo en log del CLI , el
problema es que se corta y en el telefono veo un error "500"

 Executing [2685243 at netsoluciones:1] Dial("SIP/192.168.10.1-092e00e0",
"Zap/g0/2685243|68|tT") in new stack
    -- Called g0/2685243
    -- Zap/4-1 answered SIP/192.168.10.1-092e00e0
[Aug 21 09:38:54] WARNING[3501]: chan_sip.c:1949 retrans_pkt: Maximum
retries exceeded on transmission e2f1a222eac63743 at 192.168.10.28 for
seqno 10432 (Critical Response)
[Aug 21 09:38:54] WARNING[3501]: chan_sip.c:1973 retrans_pkt: Hanging
up call e2f1a222eac63743 at 192.168.10.28 - no reply to our critical
packet.
    -- Hungup 'Zap/4-1'
  == Spawn extension (netsoluciones, 2685243, 1) exited non-zero on
'SIP/192.168.10.1-092e00e0'


en la traza del sip , veo que siempre me envía el 407 de autenticacion
y un error 500

U +2.534742 192.168.10.28:5060 -> 192.168.10.1:5060
INVITE sip:2685243 at 192.168.10.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKb72058274f5128f3
From: <sip:119 at 192.168.10.1>;tag=2ad1fbd1cac62687
To: <sip:2685243 at 192.168.10.1>
Contact: <sip:119 at 192.168.10.28:5060>
Supported: replaces, timer, path
Call-ID: a1e3e645ef330d17 at 192.168.10.28
CSeq: 7359 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 +0.001876 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=z9hG4bKb72058274f5128f3
From: <sip:119 at 192.168.10.1>;tag=2ad1fbd1cac62687
To: <sip:2685243 at 192.168.10.1>;tag=d7e793d797690be5ce5420203095040f.9e9c
Call-ID: a1e3e645ef330d17 at 192.168.10.28
CSeq: 7359 INVITE
Proxy-Authenticate: Digest realm="192.168.10.1",
nonce="48ad8e7438f62523ee6e0faf88519fa04f9889a9"
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0

U +0.000108 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 500 Server error occurred (19/SL)
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK26d182170fe295a0
From: <sip:119 at 192.168.10.1>;tag=2ad1fbd1cac62687
To: <sip:2685243 at 192.168.10.1>;tag=d7e793d797690be5ce5420203095040f.59ac
Call-ID: a1e3e645ef330d17 at 192.168.10.28
CSeq: 7360 INVITE
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0


#
U +0.000349 192.168.10.1:5070 -> 192.168.10.1:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.10.1;branch=z9hG4bKd751.0495b837.0;received=192.168.10.1
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK26d182170fe295a0
Record-Route: <sip:192.168.10.1;lr=on;ftag=2ad1fbd1cac62687>
From: <sip:119 at 192.168.10.1>;tag=2ad1fbd1cac62687
To: <sip:2685243 at 192.168.10.1>
Call-ID: a1e3e645ef330d17 at 192.168.10.28
CSeq: 7360 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2685243 at 192.168.10.1:5070>
Content-Length: 0


#
U +0.001529 192.168.10.28:5060 -> 192.168.10.1:5060
ACK sip:2685243 at 192.168.10.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK26d182170fe295a0
From: <sip:119 at 192.168.10.1>;tag=2ad1fbd1cac62687
To: <sip:2685243 at 192.168.10.1>;tag=d7e793d797690be5ce5420203095040f.59ac
Contact: <sip:119 at 192.168.10.28:5060>
Proxy-Authorization: Digest username="119", realm="192.168.10.1",
algorithm=MD5, uri="sip:2685243 at 192.168.10.1",
nonce="48ad8e7438f62523ee6e0faf88519fa04f9889a9",
response="1bb290a28d35edbd4f2daf932fdd50a9"
Call-ID: a1e3e645ef330d17 at 192.168.10.28
CSeq: 7360 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

saludoss, cualquier cosa puedo mostar el cfg de openser ..

rickygm