[asterisk-users] retransmision error con asterisk 1.4.24.1

troxlinux xserverlinux at gmail.com
Sun Apr 12 21:22:20 CDT 2009 

uff  , no me fije que envié un mensaje en español a la lista de ingles ...

I send sip log


---
Retransmitting #2 (NAT) to 192.168.10.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
CSeq: 30032 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*981 at 192.168.10.3:5070>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 3005 3005 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
t=0 0
m=audio 13584 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (NAT) to 192.168.10.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
CSeq: 30032 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*981 at 192.168.10.3:5070>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 3005 3005 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
t=0 0
m=audio 13584 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- <SIP/111-08d20da8> Playing 'vm-password' (language 'es')
Retransmitting #4 (NAT) to 192.168.10.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
CSeq: 30032 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*981 at 192.168.10.3:5070>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 3005 3005 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
t=0 0
m=audio 13584 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- <SIP/111-08d20da8> Playing 'vm-youhave' (language 'es')
Reliably Transmitting (NAT) to 192.168.10.3:5060:
OPTIONS sip:192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK63ca3652;rport
From: "asterisk" <sip:asterisk at 192.168.10.3:5070>;tag=as690b573d
To: <sip:192.168.10.3>
Contact: <sip:asterisk at 192.168.10.3:5070>
Call-ID: 36b27df370b95454445ab61d5c8c251b at 192.168.10.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 13 Apr 2009 02:20:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
twoxserver*CLI>
<--- SIP read from 192.168.10.3:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK63ca3652;rport=5070
From: "asterisk" <sip:asterisk at 192.168.10.3:5070>;tag=as690b573d
To: <sip:192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.478d
Call-ID: 36b27df370b95454445ab61d5c8c251b at 192.168.10.3
CSeq: 102 OPTIONS
Server: OpenSIPS (1.5.0-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'36b27df370b95454445ab61d5c8c251b at 192.168.10.3' Method: OPTIONS
    -- <SIP/111-08d20da8> Playing 'digits/6' (language 'es')
    -- <SIP/111-08d20da8> Playing 'vm-messages' (language 'es')
    -- <SIP/111-08d20da8> Playing 'vm-first' (language 'es')
    -- <SIP/111-08d20da8> Playing 'vm-message' (language 'es')
  == Parsing '/var/spool/asterisk/voicemail/default/111/INBOX/msg0000.txt':
Found
Retransmitting #5 (NAT) to 192.168.10.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
CSeq: 30032 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*981 at 192.168.10.3:5070>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 3005 3005 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
t=0 0
m=audio 13584 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- <SIP/111-08d20da8> Playing 'vm-received' (language 'es')
    -- <SIP/111-08d20da8> Playing 'digits/yesterday' (language 'es')
    -- <SIP/111-08d20da8> Playing 'digits/at' (language 'es')
    -- <SIP/111-08d20da8> Playing 'digits/8' (language 'es')
    -- <SIP/111-08d20da8> Playing 'digits/30' (language 'es')
Retransmitting #6 (NAT) to 192.168.10.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
CSeq: 30032 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*981 at 192.168.10.3:5070>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 3005 3005 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
t=0 0
m=audio 13584 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- <SIP/111-08d20da8> Playing 'digits/and' (language 'es')
    -- <SIP/111-08d20da8> Playing 'digits/9' (language 'es')
    -- <SIP/111-08d20da8> Playing 'digits/p-m' (language 'es')
    -- <SIP/111-08d20da8> Playing
'/var/spool/asterisk/voicemail/default/111/INBOX/msg0000' (language
'es')
[Apr 12 20:20:36] WARNING[3528]: app_voicemail.c:5619 play_message:
Playback of message
/var/spool/asterisk/voicemail/default/111/INBOX/msg0000 failed
    -- <SIP/111-08d20da8> Playing 'vm-advopts' (language 'es')
[Apr 12 20:20:38] WARNING[3140]: chan_sip.c:1976 retrans_pkt: Maximum
retries exceeded on transmission 5c4fa1fdece6f915 at 192.168.10.23 for
seqno 30032 (Critical Response) -- See doc/sip-retransmit.txt.
[Apr 12 20:20:38] WARNING[3140]: chan_sip.c:1998 retrans_pkt: Hanging
up call 5c4fa1fdece6f915 at 192.168.10.23 - no reply to our critical
packet (see doc/sip-retransmit.txt).
  == Spawn extension (netsoluciones, *981, 2) exited non-zero on
'SIP/111-08d20da8'
Really destroying SIP dialog '5c4fa1fdece6f915 at 192.168.10.23' Method: INVITE


2009/4/12 Alex Balashov <abalashov at evaristesys.com>:
> Mejor que obtengamos un packet capture para investigarlo mas.

sera un bug o algo por el estilo

saludoss

-- 
rickygm

http://gnuforever.homelinux.com

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