<div dir="ltr">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 :<br>
<br>chan_sip.c:5095 process_sdp: Unable to lookup host in c= line, 'IN IP4 192.168.1.64192.168.1.64'<br><br>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 ..<br>
<br>algo que me este faltando del realtime? ..<br><br>saludos lista<br><br>rickygm<br><br>### sip debug asterisk ####<br><br>--- (18 headers 24 lines) ---<br>Sending to <a href="http://192.168.10.1">192.168.10.1</a> : 5060 (NAT)<br>
Using INVITE request as basis request - <a href="mailto:e3b32187ee145e96@192.168.10.28">e3b32187ee145e96@192.168.10.28</a><br>Found user '119'<br>[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'<br>
<br><--- Reliably Transmitting (NAT) to <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> ---><br>SIP/2.0 488 Not acceptable here<br>Via: SIP/2.0/UDP <a href="http://192.168.10.1">192.168.10.1</a>;branch=z9hG4bK91af.827e4ad4.1;received=<a href="http://192.168.10.1">192.168.10.1</a><br>
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK1a8140c1b66271b0<br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=7dc2f363e7b65552<br>To: <<a href="mailto:sip%3A114@192.168.10.1">sip:114@192.168.10.1</a>>;tag=as5a93f32a<br>
Call-ID: <a href="mailto:e3b32187ee145e96@192.168.10.28">e3b32187ee145e96@192.168.10.28</a><br>CSeq: 25247 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>
Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog '<a href="mailto:e3b32187ee145e96@192.168.10.28">e3b32187ee145e96@192.168.10.28</a>' in 32000 ms (Method: INVITE)<br>xserver*CLI><br>
<--- SIP read from <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> ---><br>ACK <a href="http://sip:u114@192.168.10.1:5070">sip:u114@192.168.10.1:5070</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.10.1">192.168.10.1</a>;branch=z9hG4bK91af.827e4ad4.1<br>
From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=7dc2f363e7b65552<br>Call-ID: <a href="mailto:e3b32187ee145e96@192.168.10.28">e3b32187ee145e96@192.168.10.28</a><br>To: <<a href="mailto:sip%3A114@192.168.10.1">sip:114@192.168.10.1</a>>;tag=as5a93f32a<br>
CSeq: 25247 ACK<br>Max-Forwards: 70<br>User-Agent: OpenSER<br><br><br>### mi log sip Openser ####<br><br><pre>U +1.244159 <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> -> <a href="http://192.168.10.28:5060">192.168.10.28:5060</a><br>
SIP/2.0 488 Not acceptable here <br>Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKb7a454e6f1909ec6 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=d096ebe220002321 <br>
To: <<a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a>>;tag=as0cd8584b <br>Call-ID: <a href="mailto:3bb16f05ffa03a24@192.168.10.28">3bb16f05ffa03a24@192.168.10.28</a> <br>CSeq: 30320 INVITE <br>User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br> <br><br><br>#<br>U +0.172349 <a href="http://192.168.10.28:5060">192.168.10.28:5060</a> -> <a href="http://192.168.10.1:5060">192.168.10.1:5060</a><br>
INVITE <a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a> SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKf306872728753147 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=45d17c11e5a1f791 <br>
To: <<a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a>> <br>Contact: <<a href="http://sip:119@192.168.10.28:5060">sip:119@192.168.10.28:5060</a>> <br>Supported: replaces, timer, path <br>Call-ID: <a href="mailto:3bd1333498b35042@192.168.10.28">3bd1333498b35042@192.168.10.28</a> <br>
CSeq: 26761 INVITE <br>User-Agent: Grandstream GXV3000 <a href="http://1.1.3.14">1.1.3.14</a> <br>Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK <br>Content-Type: application/sdp <br>
Content-Length: 545 <br> <br>v=0 <br>o=119 8000 8000 IN IP4 <a href="http://192.168.10.28">192.168.10.28</a> <br>s=SIP Call <br>c=IN IP4 <a href="http://192.168.10.28">192.168.10.28</a> <br>t=0 0 <br>m=audio 5004 RTP/AVP 0 18 4 3 2 101 <br>
a=sendrecv <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:4 G723/8000 <br>a=rtpmap:3 GSM/8000 <br>a=rtpmap:2 G726-32/8000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11 <br>m=video 5006 RTP/AVP 99 34 <br>
a=sendrecv <br>a=rtpmap:99 H264/90000 <br>a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4BryA= <br>a=rtpmap:34 H263/90000 <br>a=fmtp:34 CIF=2 MaxBR=1280 <br>a=framerate:20 <br>
<br>#<br>U +0.000232 <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> -> <a href="http://192.168.10.28:5060">192.168.10.28:5060</a><br>SIP/2.0 407 Proxy Authentication Required <br>Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKf306872728753147;rport=5060 <br>
From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=45d17c11e5a1f791 <br>To: <<a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a>>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4555 <br>
Call-ID: <a href="mailto:3bd1333498b35042@192.168.10.28">3bd1333498b35042@192.168.10.28</a> <br>CSeq: 26761 INVITE <br>Proxy-Authenticate: Digest realm="<a href="http://192.168.10.1">192.168.10.1</a>", nonce="48f427bd23837102189868c302984a0314f3ac5e" <br>
Server: OpenSER (1.3.2-notls (i386/linux)) <br>Content-Length: 0 <br> <br><br>#<br>U +0.001676 <a href="http://192.168.10.28:5060">192.168.10.28:5060</a> -> <a href="http://192.168.10.1:5060">192.168.10.1:5060</a><br>ACK <a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a> SIP/2.0 <br>
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKf306872728753147 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=45d17c11e5a1f791 <br>To: <<a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a>>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4555 <br>
Contact: <<a href="http://sip:119@192.168.10.28:5060">sip:119@192.168.10.28:5060</a>> <br>Call-ID: <a href="mailto:3bd1333498b35042@192.168.10.28">3bd1333498b35042@192.168.10.28</a> <br>CSeq: 26761 ACK <br>User-Agent: Grandstream GXV3000 <a href="http://1.1.3.14">1.1.3.14</a> <br>
Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK <br>Content-Length: 0 <br> <br><br>#<br>U +0.001321 <a href="http://192.168.10.28:5060">192.168.10.28:5060</a> -> <a href="http://192.168.10.1:5060">192.168.10.1:5060</a><br>
INVITE <a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a> SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK5bb5e1738e335481 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=45d17c11e5a1f791 <br>
To: <<a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a>> <br>Contact: <<a href="http://sip:119@192.168.10.28:5060">sip:119@192.168.10.28:5060</a>> <br>Supported: replaces, timer, path <br>Proxy-Authorization: Digest username="119", realm="<a href="http://192.168.10.1">192.168.10.1</a>", algorithm=MD5, uri="<a href="mailto:sip%3A111@192.168.10.1">sip:111@192.168.10.1</a>", nonce="48f427bd23837102189868c302984a0314f3ac5e", response="d5d20ad6310dea0460b3f4ca8b020e27" <br>
Call-ID: <a href="mailto:3bd1333498b35042@192.168.10.28">3bd1333498b35042@192.168.10.28</a> <br>CSeq: 26762 INVITE <br>User-Agent: Grandstream GXV3000 <a href="http://1.1.3.14">1.1.3.14</a> <br>Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK <br>
Content-Type: application/sdp <br>Content-Length: 545 <br> <br>v=0 <br>o=119 8000 8001 IN IP4 <a href="http://192.168.10.28">192.168.10.28</a> <br>s=SIP Call <br>c=IN IP4 <a href="http://192.168.10.28">192.168.10.28</a> <br>
t=0 0 <br>m=audio 5004 RTP/AVP 0 18 4 3 2 101 <br>a=sendrecv <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:4 G723/8000 <br>a=rtpmap:3 GSM/8000 <br>a=rtpmap:2 G726-32/8000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>
a=fmtp:101 0-11 <br>m=video 5006 RTP/AVP 99 34 <br>a=sendrecv <br>a=rtpmap:99 H264/90000 <br>a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4BryA= <br>a=rtpmap:34 H263/90000 <br>
a=fmtp:34 CIF=2 MaxBR=1280 <br>a=framerate:20 <br></pre><br>
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