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Hola, tengo problemas de audio con mi pbx, a mi si me escuchan pero yo no los escucho, las llamadas entre extensiones si se escuchan sin problema. adjunto lo que me manda el debug.
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:No
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: public
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: es
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
RTCP Multiplexing: No
----
== Using SIP RTP CoS mark 5
-- Executing [56581111@contextoLADAEtsai:1] NoOp("SIP/ayavillop-0000000c", "Llamada a numero local") in new stack
-- Executing [56581111@contextoLADAEtsai:2] Set("SIP/ayavillop-0000000c", "CALLERID(all)="Etsai" <525575866660>") in new stack
-- Executing [56581111@contextoLADAEtsai:3] Dial("SIP/ayavillop-0000000c", "SIP/ipcallsOut/525556581111,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/ipcallsOut/525556581111
-- SIP/ipcallsOut-0000000d is making progress passing it to SIP/ayavillop-0000000c
-- SIP/ipcallsOut-0000000d redirecting info has changed, passing it to SIP/ayavillop-0000000c
-- SIP/ipcallsOut-0000000d is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [56581111@contextoLADAEtsai:4] Hangup("SIP/ayavillop-0000000c", "") in new stack
== Spawn extension (contextoLADAEtsai, 56581111, 4) exited non-zero on 'SIP/ayavillop-0000000c'
== Using SIP RTP CoS mark 5
-- Executing [56581111@contextoLADAEtsai:1] NoOp("SIP/ayavillop-0000000e", "Llamada a numero local") in new stack
-- Executing [56581111@contextoLADAEtsai:2] Set("SIP/ayavillop-0000000e", "CALLERID(all)="Etsai" <525575866660>") in new stack
-- Executing [56581111@contextoLADAEtsai:3] Dial("SIP/ayavillop-0000000e", "SIP/ipcallsOut/525556581111,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/ipcallsOut/525556581111
-- SIP/ipcallsOut-0000000f is ringing
-- SIP/ipcallsOut-0000000f is making progress passing it to SIP/ayavillop-0000000e
-- SIP/ipcallsOut-0000000f answered SIP/ayavillop-0000000e
-- Channel SIP/ipcallsOut-0000000f joined 'simple_bridge' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
-- Channel SIP/ayavillop-0000000e joined 'simple_bridge' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
-- Channel SIP/ayavillop-0000000e left 'native_rtp' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
== Spawn extension (contextoLADAEtsai, 56581111, 3) exited non-zero on 'SIP/ayavillop-0000000e'
-- Channel SIP/ipcallsOut-0000000f left 'native_rtp' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
server125*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:10.8.22.14:49930 --->
<------------->
<--- SIP read from UDP:10.8.22.10:5060 --->
<------------->
<--- SIP read from UDP:10.8.22.14:49930 --->
INVITE sip:56581111@10.8.22.1 SIP/2.0
Via: SIP/2.0/UDP 10.8.22.14:49930;rport;branch=z9hG4bKPj6ca870eae27 e4bb88b7acca34c847dcf
Max-Forwards: 70
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>
Contact: "Enrique" <sip:ayavillop@10.8.22.14:49930;ob>
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16882 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.16.9
Content-Type: application/sdp
Content-Length: 365
v=0
o=- 3727514019 3727514019 IN IP4 192.168.0.5
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 18 0 8 3 101
c=IN IP4 192.168.0.5
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.0.5
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (15 headers 18 lines) ---
Sending to 10.8.22.14:49930 (NAT)
Sending to 10.8.22.14:49930 (NAT)
Using INVITE request as basis request - acad433ed1454e58b2eeb76bbcbc3b5d
Found peer 'ayavillop' for 'ayavillop' from 10.8.22.14:49930
<--- Reliably Transmitting (no NAT) to 10.8.22.14:49930 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.22.14:49930;branch=z9hG4bKPj6ca870eae27e4bb88 b7acca34c847dcf;received=10.8.22.14;rport=49930
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>;tag=as01f097d0
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16882 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f416278"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'acad433ed1454e58b2eeb76bbcbc3b5d' in 9536 ms (Method: INVITE)
<--- SIP read from UDP:10.8.22.14:49930 --->
ACK sip:56581111@10.8.22.1 SIP/2.0
Via: SIP/2.0/UDP 10.8.22.14:49930;rport;branch=z9hG4bKPj6ca870eae27 e4bb88b7acca34c847dcf
Max-Forwards: 70
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>;tag=as01f097d0
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16882 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.8.22.14:49930 --->
INVITE sip:56581111@10.8.22.1 SIP/2.0
Via: SIP/2.0/UDP 10.8.22.14:49930;rport;branch=z9hG4bKPj2a21b03645a a4195b96f6f0126943e7f
Max-Forwards: 70
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>
Contact: "Enrique" <sip:ayavillop@10.8.22.14:49930;ob>
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16883 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.16.9
Authorization: Digest username="ayavillop", realm="asterisk", nonce="3f416278", uri="sip:56581111@10.8.22.1", response="1a8cedab77615a8d2909215dd97a4dfd", algorithm=MD5
Content-Type: application/sdp
Content-Length: 365
v=0
o=- 3727514019 3727514019 IN IP4 192.168.0.5
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 18 0 8 3 101
c=IN IP4 192.168.0.5
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.0.5
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (16 headers 18 lines) ---
Sending to 10.8.22.14:49930 (no NAT)
Using INVITE request as basis request - acad433ed1454e58b2eeb76bbcbc3b5d
Found peer 'ayavillop' for 'ayavillop' from 10.8.22.14:49930
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|ulaw|alaw|gsm), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (g729|ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.5:4006
Looking for 56581111 in contextoLADAEtsai (domain 10.8.22.1)
sip_route_dump: route/path hop: <sip:ayavillop@10.8.22.14:49930;ob>
<--- Transmitting (no NAT) to 10.8.22.14:49930 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.22.14:49930;branch=z9hG4bKPj2a21b03645aa4195b 96f6f0126943e7f;received=10.8.22.14;rport=49930
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16883 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:56581111@10.8.22.1:5060>
Content-Length: 0
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:No
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: public
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: es
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
RTCP Multiplexing: No
----
== Using SIP RTP CoS mark 5
-- Executing [56581111@contextoLADAEtsai:1] NoOp("SIP/ayavillop-0000000c", "Llamada a numero local") in new stack
-- Executing [56581111@contextoLADAEtsai:2] Set("SIP/ayavillop-0000000c", "CALLERID(all)="Etsai" <525575866660>") in new stack
-- Executing [56581111@contextoLADAEtsai:3] Dial("SIP/ayavillop-0000000c", "SIP/ipcallsOut/525556581111,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/ipcallsOut/525556581111
-- SIP/ipcallsOut-0000000d is making progress passing it to SIP/ayavillop-0000000c
-- SIP/ipcallsOut-0000000d redirecting info has changed, passing it to SIP/ayavillop-0000000c
-- SIP/ipcallsOut-0000000d is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [56581111@contextoLADAEtsai:4] Hangup("SIP/ayavillop-0000000c", "") in new stack
== Spawn extension (contextoLADAEtsai, 56581111, 4) exited non-zero on 'SIP/ayavillop-0000000c'
== Using SIP RTP CoS mark 5
-- Executing [56581111@contextoLADAEtsai:1] NoOp("SIP/ayavillop-0000000e", "Llamada a numero local") in new stack
-- Executing [56581111@contextoLADAEtsai:2] Set("SIP/ayavillop-0000000e", "CALLERID(all)="Etsai" <525575866660>") in new stack
-- Executing [56581111@contextoLADAEtsai:3] Dial("SIP/ayavillop-0000000e", "SIP/ipcallsOut/525556581111,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/ipcallsOut/525556581111
-- SIP/ipcallsOut-0000000f is ringing
-- SIP/ipcallsOut-0000000f is making progress passing it to SIP/ayavillop-0000000e
-- SIP/ipcallsOut-0000000f answered SIP/ayavillop-0000000e
-- Channel SIP/ipcallsOut-0000000f joined 'simple_bridge' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
-- Channel SIP/ayavillop-0000000e joined 'simple_bridge' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
-- Channel SIP/ayavillop-0000000e left 'native_rtp' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
== Spawn extension (contextoLADAEtsai, 56581111, 3) exited non-zero on 'SIP/ayavillop-0000000e'
-- Channel SIP/ipcallsOut-0000000f left 'native_rtp' basic-bridge <7ea03f36-dcf3-4594-b71d-57312ad1ce45>
server125*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:10.8.22.14:49930 --->
<------------->
<--- SIP read from UDP:10.8.22.10:5060 --->
<------------->
<--- SIP read from UDP:10.8.22.14:49930 --->
INVITE sip:56581111@10.8.22.1 SIP/2.0
Via: SIP/2.0/UDP 10.8.22.14:49930;rport;branch=z9hG4bKPj6ca870eae27 e4bb88b7acca34c847dcf
Max-Forwards: 70
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>
Contact: "Enrique" <sip:ayavillop@10.8.22.14:49930;ob>
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16882 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.16.9
Content-Type: application/sdp
Content-Length: 365
v=0
o=- 3727514019 3727514019 IN IP4 192.168.0.5
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 18 0 8 3 101
c=IN IP4 192.168.0.5
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.0.5
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (15 headers 18 lines) ---
Sending to 10.8.22.14:49930 (NAT)
Sending to 10.8.22.14:49930 (NAT)
Using INVITE request as basis request - acad433ed1454e58b2eeb76bbcbc3b5d
Found peer 'ayavillop' for 'ayavillop' from 10.8.22.14:49930
<--- Reliably Transmitting (no NAT) to 10.8.22.14:49930 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.22.14:49930;branch=z9hG4bKPj6ca870eae27e4bb88 b7acca34c847dcf;received=10.8.22.14;rport=49930
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>;tag=as01f097d0
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16882 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f416278"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'acad433ed1454e58b2eeb76bbcbc3b5d' in 9536 ms (Method: INVITE)
<--- SIP read from UDP:10.8.22.14:49930 --->
ACK sip:56581111@10.8.22.1 SIP/2.0
Via: SIP/2.0/UDP 10.8.22.14:49930;rport;branch=z9hG4bKPj6ca870eae27 e4bb88b7acca34c847dcf
Max-Forwards: 70
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>;tag=as01f097d0
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16882 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.8.22.14:49930 --->
INVITE sip:56581111@10.8.22.1 SIP/2.0
Via: SIP/2.0/UDP 10.8.22.14:49930;rport;branch=z9hG4bKPj2a21b03645a a4195b96f6f0126943e7f
Max-Forwards: 70
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>
Contact: "Enrique" <sip:ayavillop@10.8.22.14:49930;ob>
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16883 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.16.9
Authorization: Digest username="ayavillop", realm="asterisk", nonce="3f416278", uri="sip:56581111@10.8.22.1", response="1a8cedab77615a8d2909215dd97a4dfd", algorithm=MD5
Content-Type: application/sdp
Content-Length: 365
v=0
o=- 3727514019 3727514019 IN IP4 192.168.0.5
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 18 0 8 3 101
c=IN IP4 192.168.0.5
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.0.5
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (16 headers 18 lines) ---
Sending to 10.8.22.14:49930 (no NAT)
Using INVITE request as basis request - acad433ed1454e58b2eeb76bbcbc3b5d
Found peer 'ayavillop' for 'ayavillop' from 10.8.22.14:49930
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|ulaw|alaw|gsm), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (g729|ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.5:4006
Looking for 56581111 in contextoLADAEtsai (domain 10.8.22.1)
sip_route_dump: route/path hop: <sip:ayavillop@10.8.22.14:49930;ob>
<--- Transmitting (no NAT) to 10.8.22.14:49930 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.22.14:49930;branch=z9hG4bKPj2a21b03645aa4195b 96f6f0126943e7f;received=10.8.22.14;rport=49930
From: "Enrique" <sip:ayavillop@10.8.22.1>;tag=d424e458dc504948a68c f9996cbac4c0
To: <sip:56581111@10.8.22.1>
Call-ID: acad433ed1454e58b2eeb76bbcbc3b5d
CSeq: 16883 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:56581111@10.8.22.1:5060>
Content-Length: 0