No voice

Asterisk Asterisk Support
1

Hi,

I can make call but no one hear anything.
I added add a Playback(hello-world) exten in dialplan but still I can not hear anything?

Any idea?
Thanks

2

This happend in your LAN or outside your LAN, can you put the CLI output?

3

This happens in our LAN and makind outside calls…
The log:

<— SIP read from 192.168.2.214:58942 —>
INVITE sip:2001@192.168.2.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-9871e43d0f207940-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@192.168.2.214:58942
To: "2001"sip:2001@192.168.2.76
From: "2000"sip:2000@192.168.2.76;tag=e54b8716
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 584

v=0
o=- 9 2 IN IP4 192.168.2.214
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.214
t=0 0
m=audio 17570 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101
a=alt:1 3 : DtYSKIGc IY8URlRz 192.168.5.36 17570
a=alt:2 2 : SOS92xW/ x6zoIJFS 192.168.2.214 17570
a=alt:3 1 : 3lXLlLHR 8JIBfFBg 192.168.56.1 17570
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:102 L16/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (12 headers 20 lines) —
Sending to 192.168.2.214 : 58942 (no NAT)
Using INVITE request as basis request - MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.

<— Reliably Transmitting (no NAT) to 192.168.2.214:58942 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-9871e43d0f207940-1—d8754z-;received=192.168.2.214;rport=58942
From: "2000"sip:2000@192.168.2.76;tag=e54b8716
To: "2001"sip:2001@192.168.2.76;tag=as5012a815
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e7dd20e"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.’ in 32000 ms (Method: INVITE)
Found user ‘2000’

<— SIP read from 192.168.2.214:58942 —>
ACK sip:2001@192.168.2.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-9871e43d0f207940-1—d8754z-;rport
To: "2001"sip:2001@192.168.2.76;tag=as5012a815
From: "2000"sip:2000@192.168.2.76;tag=e54b8716
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from 192.168.2.214:58942 —>
INVITE sip:2001@192.168.2.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-2923a525522f9524-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@192.168.2.214:58942
To: "2001"sip:2001@192.168.2.76
From: “2000"sip:2000@192.168.2.76;tag=e54b8716
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username=“2000”,realm=“asterisk”,nonce=“4e7dd20e”,uri="sip:2001@192.168.2.76”,response=“7500d12fc728c9a861b4019db04ccdc8”,algorithm=MD5
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 584

v=0
o=- 9 2 IN IP4 192.168.2.214
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.214
t=0 0
m=audio 17570 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101
a=alt:1 3 : DtYSKIGc IY8URlRz 192.168.5.36 17570
a=alt:2 2 : SOS92xW/ x6zoIJFS 192.168.2.214 17570
a=alt:3 1 : 3lXLlLHR 8JIBfFBg 192.168.56.1 17570
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:102 L16/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (13 headers 20 lines) —
Sending to 192.168.2.214 : 58942 (NAT)
Using INVITE request as basis request - MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
Found user '2000’
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 102
Found RTP audio format 3
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.214:17570
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found audio description format SPEEX for ID 97
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format L16 for ID 102
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x66e (gsm|ulaw|alaw|adpcm|slin|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.214:17570
Looking for 2001 in superonline (domain 192.168.2.76)
list_route: hop: sip:2000@192.168.2.214:58942

<— Transmitting (NAT) to 192.168.2.214:58942 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-2923a525522f9524-1—d8754z-;received=192.168.2.214;rport=58942
From: "2000"sip:2000@192.168.2.76;tag=e54b8716
To: "2001"sip:2001@192.168.2.76
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:2001@192.168.2.76
Content-Length: 0

<------------>
– Executing [2001@superonline:1] Dial(“SIP/2000-b00061d0”, “SIP/2001|20”) in new stack
Audio is at 192.168.2.76 port 14380
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.220:7498:
INVITE sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK374c67f8;rport
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
Contact: sip:2000@192.168.2.76
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 10 Dec 2009 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 3620 3620 IN IP4 192.168.2.76
s=session
c=IN IP4 192.168.2.76
t=0 0
m=audio 14380 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called 2001

<— SIP read from 192.168.2.220:7498 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK374c67f8;rport=5060
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from 192.168.2.220:7498 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK374c67f8;rport=5060
Contact: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
From: "2000"sip:2000@192.168.2.76;tag=as3716836c
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 102 INVITE
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0

<------------->
— (9 headers 0 lines) —
– SIP/2001-0af7e1c0 is ringing

<— Transmitting (NAT) to 192.168.2.214:58942 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-2923a525522f9524-1—d8754z-;received=192.168.2.214;rport=58942
From: "2000"sip:2000@192.168.2.76;tag=e54b8716
To: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:2001@192.168.2.76
Content-Length: 0

<------------>

<— SIP read from 192.168.2.214:58942 —>

<------------->

<— SIP read from 192.168.2.220:7498 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK374c67f8;rport=5060
Contact: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
From: "2000"sip:2000@192.168.2.76;tag=as3716836c
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 189

v=0
o=- 5 2 IN IP4 192.168.2.220
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.220
t=0 0
m=audio 46186 RTP/AVP 0 3 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (11 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.220:46186
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.220:46186
list_route: hop: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
set_destination: Parsing sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce for address/port to send to
set_destination: set destination to 192.168.2.220, port 7498
Transmitting (no NAT) to 192.168.2.220:7498:
ACK sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK389bedbf;rport
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
Contact: sip:2000@192.168.2.76
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/2001-0af7e1c0 answered SIP/2000-b00061d0

Audio is at 192.168.2.76 port 14180
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.2.214:58942 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-2923a525522f9524-1—d8754z-;received=192.168.2.214;rport=58942
From: "2000"sip:2000@192.168.2.76;tag=e54b8716
To: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:2001@192.168.2.76
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 3620 3620 IN IP4 192.168.2.76
s=session
c=IN IP4 192.168.2.76
t=0 0
m=audio 14180 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Native bridging SIP/2000-b00061d0 and SIP/2001-0af7e1c0
set_destination: Parsing sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce for address/port to send to
set_destination: set destination to 192.168.2.220, port 7498
Audio is at 192.168.2.76 port 14380
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.220:7498:
INVITE sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK2d2cb465;rport
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
Contact: sip:2000@192.168.2.76
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 3620 3621 IN IP4 192.168.2.214
s=session
c=IN IP4 192.168.2.214
t=0 0
m=audio 17570 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from 192.168.2.220:7498 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK2d2cb465;rport=5060
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 103 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from 192.168.2.214:58942 —>
ACK sip:2001@192.168.2.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-1076e505f0754703-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@192.168.2.214:58942
To: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
From: “2000"sip:2000@192.168.2.76;tag=e54b8716
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 2 ACK
Proxy-Authorization: Digest username=“2000”,realm=“asterisk”,nonce=“4e7dd20e”,uri="sip:2001@192.168.2.76”,response=“7500d12fc728c9a861b4019db04ccdc8”,algorithm=MD5
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0

<------------->
— (11 headers 0 lines) —
set_destination: Parsing sip:2000@192.168.2.214:58942 for address/port to send to
set_destination: set destination to 192.168.2.214, port 58942
Audio is at 192.168.2.76 port 14180
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.2.214:58942:
INVITE sip:2000@192.168.2.214:58942 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK033a8483;rport
From: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
To: "2000"sip:2000@192.168.2.76;tag=e54b8716
Contact: sip:2001@192.168.2.76
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 3620 3621 IN IP4 192.168.2.220
s=session
c=IN IP4 192.168.2.220
t=0 0
m=audio 46186 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from 192.168.2.220:7498 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK2d2cb465;rport=5060
Contact: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
From: "2000"sip:2000@192.168.2.76;tag=as3716836c
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 189

v=0
o=- 5 2 IN IP4 192.168.2.220
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.220
t=0 0
m=audio 46186 RTP/AVP 0 3 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (11 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.220:46186
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.220:46186
set_destination: Parsing sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce for address/port to send to
set_destination: set destination to 192.168.2.220, port 7498
Transmitting (no NAT) to 192.168.2.220:7498:
ACK sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK067cbd72;rport
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
Contact: sip:2000@192.168.2.76
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<— SIP read from 192.168.2.214:58942 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK033a8483;rport=5060
Contact: sip:2000@192.168.2.214:58942
To: "2000"sip:2000@192.168.2.76;tag=e54b8716
From: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 189

v=0
o=- 9 3 IN IP4 192.168.2.214
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.214
t=0 0
m=audio 17570 RTP/AVP 3 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (11 headers 9 lines) —
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.214:17570
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.214:17570
set_destination: Parsing sip:2000@192.168.2.214:58942 for address/port to send to
set_destination: set destination to 192.168.2.214, port 58942
Transmitting (NAT) to 192.168.2.214:58942:
ACK sip:2000@192.168.2.214:58942 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK4b20954a;rport
From: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
To: "2000"sip:2000@192.168.2.76;tag=e54b8716
Contact: sip:2001@192.168.2.76
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<— SIP read from 192.168.2.214:58942 —>
BYE sip:2001@192.168.2.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-d95d7e0be27e9656-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@192.168.2.214:58942
To: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
From: “2000"sip:2000@192.168.2.76;tag=e54b8716
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 3 BYE
Proxy-Authorization: Digest username=“2000”,realm=“asterisk”,nonce=“4e7dd20e”,uri="sip:2001@192.168.2.76”,response=“fae9e4c7513f85d18070e22aac4ae0eb”,algorithm=MD5
User-Agent: X-Lite release 1103k stamp 53621
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.2.214 : 58942 (NAT)

<— Transmitting (NAT) to 192.168.2.214:58942 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-d95d7e0be27e9656-1—d8754z-;received=192.168.2.214;rport=58942
From: "2000"sip:2000@192.168.2.76;tag=e54b8716
To: "2001"sip:2001@192.168.2.76;tag=as7a35b30d
Call-ID: MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
set_destination: Parsing sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce for address/port to send to
set_destination: set destination to 192.168.2.220, port 7498
Audio is at 192.168.2.76 port 14380
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.220:7498:
INVITE sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK19a8b15a;rport
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
Contact: sip:2000@192.168.2.76
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 3620 3622 IN IP4 192.168.2.76
s=session
c=IN IP4 192.168.2.76
t=0 0
m=audio 14380 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Scheduling destruction of SIP dialog ‘7e0805ae4c98e861369db5223a8cbd67@192.168.2.76’ in 32000 ms (Method: INVITE)
== Spawn extension (superonline, 2001, 1) exited non-zero on ‘SIP/2000-b00061d0’

<— SIP read from 192.168.2.220:7498 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK19a8b15a;rport=5060
Contact: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
From: "2000"sip:2000@192.168.2.76;tag=as3716836c
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 104 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 189

v=0
o=- 5 2 IN IP4 192.168.2.220
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.220
t=0 0
m=audio 46186 RTP/AVP 0 3 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (11 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.220:46186
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.220:46186
set_destination: Parsing sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce for address/port to send to
set_destination: set destination to 192.168.2.220, port 7498
Transmitting (no NAT) to 192.168.2.220:7498:
ACK sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK2c044493;rport
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
Contact: sip:2000@192.168.2.76
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

set_destination: Parsing sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce for address/port to send to
set_destination: set destination to 192.168.2.220, port 7498
Reliably Transmitting (no NAT) to 192.168.2.220:7498:
BYE sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce SIP/2.0
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK27decdf7;rport
From: “2000” sip:2000@192.168.2.76;tag=as3716836c
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Scheduling destruction of SIP dialog ‘7e0805ae4c98e861369db5223a8cbd67@192.168.2.76’ in 32000 ms (Method: INVITE)
Really destroying SIP dialog ‘MjZhNmJlMjY3YmUwZjA0NjI4NDhmYmU1M2U1MDIyMGI.’ Method: BYE

<— SIP read from 192.168.2.220:7498 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK27decdf7;rport=5060
Contact: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce
To: sip:2001@192.168.2.220:7498;rinstance=222ca398a1881bce;tag=e52cfc4a
From: "2000"sip:2000@192.168.2.76;tag=as3716836c
Call-ID: 7e0805ae4c98e861369db5223a8cbd67@192.168.2.76
CSeq: 105 BYE
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘7e0805ae4c98e861369db5223a8cbd67@192.168.2.76’ Method: INVITE

<— SIP read from 192.168.2.220:7498 —>

<------------->

<— SIP read from 192.168.2.214:58942 —>

<------------->

<— SIP read from 192.168.2.214:58942 —>
SUBSCRIBE sip:2000@192.168.2.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-0270510113776438-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@192.168.2.214:58942
To: "2000"sip:2000@192.168.2.76
From: "2000"sip:2000@192.168.2.76;tag=4d570464
Call-ID: ZjdmODJkZWYwNGVmMjQ5NmUzOTQwZjU2ZDQ1OWIwZjc.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1103k stamp 53621
Event: message-summary
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Creating new subscription
Sending to 192.168.2.214 : 58942 (no NAT)
list_route: hop: sip:2000@192.168.2.214:58942
Found peer ‘2000’

<— Transmitting (no NAT) to 192.168.2.214:58942 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-0270510113776438-1—d8754z-;received=192.168.2.214;rport=58942
From: "2000"sip:2000@192.168.2.76;tag=4d570464
To: "2000"sip:2000@192.168.2.76;tag=as3f68f42b
Call-ID: ZjdmODJkZWYwNGVmMjQ5NmUzOTQwZjU2ZDQ1OWIwZjc.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="22a3ae78"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZjdmODJkZWYwNGVmMjQ5NmUzOTQwZjU2ZDQ1OWIwZjc.’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from 192.168.2.214:58942 —>
SUBSCRIBE sip:2000@192.168.2.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-c57063797107bc36-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@192.168.2.214:58942
To: "2000"sip:2000@192.168.2.76
From: “2000"sip:2000@192.168.2.76;tag=4d570464
Call-ID: ZjdmODJkZWYwNGVmMjQ5NmUzOTQwZjU2ZDQ1OWIwZjc.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username=“2000”,realm=“asterisk”,nonce=“22a3ae78”,uri="sip:2000@192.168.2.76”,response=“c81b039b9510ea7ab6b2cb06b1e950fc”,algorithm=MD5
Event: message-summary
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.2.214 : 58942 (NAT)
Found peer ‘2000’

<— Transmitting (NAT) to 192.168.2.214:58942 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.2.214:58942;branch=z9hG4bK-d8754z-c57063797107bc36-1—d8754z-;received=192.168.2.214;rport=58942
From: "2000"sip:2000@192.168.2.76;tag=4d570464
To: "2000"sip:2000@192.168.2.76;tag=as3f68f42b
Call-ID: ZjdmODJkZWYwNGVmMjQ5NmUzOTQwZjU2ZDQ1OWIwZjc.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

4

How is your router/address translator configured?

What is in you sip.conf, in particular the settings of: externhost, externip, localnet, canreinvite, and nat?

5

Hi,

This is my sip.conf setting.

[mysip]
port=5060
type=peer
context=mycontext
host=111.111.11.11
fromdomain=111.111.11.11
canreinvite=yes
disallow=all
allow=all
allow=g729

externhost, externip, localnet is closed with semi-colon.

6

As your trace shows non-public addresses, you are obviously in a NAT situation. You will need to configure at least some of the parameters, except for canreinvite. I missed one of them, which was stunaddr. The details depend on the answer to my first question. You are likely to have to change canreinvite, as well, although that also depends on the answer to the first question.

Try starting from here: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

7

I’m behind a NAT.

We work with a telecom UMTH company.
And I’m trying to make calls over internet(ADSL or cable modem) with them.
And I gave them our internet IP and they gave permission to our IP.
So does it related with this?
Because I didn’t figure out what change I have to do…

I set NAT=yes in sip.conf but nothing changed…
I still can make cals but no sounds…

Thanks

8

I didn’t say you needed to set nat. I said it was one of the things you might need to set. As the VOIP-NET article says, there are many possible NAT configurations and you need to understand yours before you can set things up properly. I suspect nat applies when Asterisk is outside the NAT and the phone is inside.

9

Following the tips&links provided by david55 you can resolve your problem. One test to be sure that is a NAT problem, put your server in dmz zone. If you hear your prompts you need to set the coorect parameters for your server in your firewall.

Do you have 1 dsl connection or 2 in the same router?

10

I have 1 ADSL modem.
I added asterisk IP to DMZ.
I can make calls and can hear voice with using x-lite.

But when I try with AMI it plays beep…

Thanks…