[Asterisk-Users] Why can't I hear audio?

Michael D Schelin mike at shelcomm.com
Mon Apr 25 10:34:51 MST 2005 

Hi Everybody can someone tell me why I can hear audio?  My call is to my
proxie which is directing it to my Asterisk box.  The Voice mail is
playing but I think its playing to my proxie.

the phone is on 198.31.185.246:63257

Here is from the sip debug.  Thanks



Sip read:
INVITE sip:9009 at 208.41.254.125 SIP/2.0
Via: SIP/2.0/UDP 208.41.254.119:5060;
branch=z9hG4bKAClkQjtHCQANCAAAAAAAALBn5V+1x2CxseR9SzOu84U_
Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4
From: "Shelcomm call forwarding test"
<sip:6262769011 at 208.41.254.119;user=phone>;tag=100c9f35ec6f09a2
To: <sip:9009 at 208.41.254.119;user=phone>
Contact: <sip:6262769011 at 198.31.185.246:63257;user=phone>
Supported: replaces
Proxy-Authorization: DIGEST username="6262769011 at sip.shelcomm.com",
realm="sip.shelcomm.com", algorithm=MD5,
uri="sip:9009 at 208.41.254.119;user=phone", qop=auth, nc=00000001,
cnonce="1a605453cf8a557d", nonce="62JsQimn5xTAMoKDHL+EbAkTRGg=",
response="874d55e7960ad550b78bb1d8660faf69"
Call-ID: fe67fc663fa36479 at 192.168.1.124
CSeq: 55676 INVITE
User-Agent: Grandstream BT100 1.0.5.23
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 338
Record-Route: <sip:208.41.254.119;lr;hash=sipd-0-2-2>
asterisk1*CLI>
v=0
o=6262769011 8000 8001 IN IP4 198.31.185.246
s=SIP Call
c=IN IP4 198.31.185.246
t=0 0
m=audio 63268 RTP/AVP 0 4 9 15 2 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:15 G728/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

16 headers, 15 lines
Using latest request as basis request
Sending to 208.41.254.119 : 5060 (non-NAT)
Found no matching peer or user for '208.41.254.119:5060'
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 9
Found RTP audio format 15
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 198.31.185.246:63268
Found description format PCMU
Found description format G723
Found description format G722
Found description format G728
Found description format G726-32
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x115
(g723|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 9009 in from-sip-external
list_route: hop: <sip:208.41.254.119;lr;hash=sipd-0-2-2>
list_route: hop: <sip:6262769011 at 198.31.185.246:63257;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.41.254.119:5060;
branch=z9hG4bKAClkQjtHCQANCAAAAAAAALBn5V+1x2CxseR9SzOu84U_
Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4
From: "Shelcomm call forwarding test"
<sip:6262769011 at 208.41.254.119;user=phone>;tag=100c9f35ec6f09a2
To: <sip:9009 at 208.41.254.119;user=phone>;tag=as59b09f62
Call-ID: fe67fc663fa36479 at 192.168.1.124
CSeq: 55676 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9009 at 208.41.254.125>
Content-Length: 0


to 208.41.254.119:5060
    -- Executing VoiceMail("SIP/208.41.254.119-089aef50", "9009") in new
stack
We're at 208.41.254.125 port 13630
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.41.254.119:5060;
branch=z9hG4bKAClkQjtHCQANCAAAAAAAALBn5V+1x2CxseR9SzOu84U_
Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4
Record-Route: <sip:208.41.254.119;lr;hash=sipd-0-2-2>
From: "Shelcomm call forwarding test"
<sip:6262769011 at 208.41.254.119;user=phone>;tag=100c9f35ec6f09a2
To: <sip:9009 at 208.41.254.119;user=phone>;tag=as59b09f62
Call-ID: fe67fc663fa36479 at 192.168.1.124
CSeq: 55676 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9009 at 208.41.254.125>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2330 2330 IN IP4 208.41.254.125
s=session
c=IN IP4 208.41.254.125
t=0 0
m=audio 13630 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

to 208.41.254.119:5060
    -- Playing 'vm-intro' (language 'en')
asterisk1*CLI>

Sip read:
ACK sip:9009 at 208.41.254.125 SIP/2.0
Via: SIP/2.0/UDP 208.41.254.119:5060;
branch=z9hG4bKAClkQjtHCQAOCAAAAAAAACEtoOY6oOebox7ZBwoRRiY_
Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bK2e698f7c1c332d46
From: "Shelcomm call forwarding test"
<sip:6262769011 at 208.41.254.119;user=phone>;tag=100c9f35ec6f09a2
To: <sip:9009 at 208.41.254.119;user=phone>;tag=as59b09f62
Contact: <sip:6262769011 at 198.31.185.246:63257;user=phone>
Proxy-Authorization: DIGEST username="6262769011 at sip.shelcomm.com",
realm="sip.shelcomm.com", algorithm=MD5, uri="sip:9009 at 208.41.254.125",
qop=auth, nc=00000002, cnonce="b85d4240018f156a",
nonce="62JsQimn5xTAMoKDHL+EbAkTRGg=",
response="4030f97656e76c9bffecee6942efbfcc"
Call-ID: fe67fc663fa36479 at 192.168.1.124
CSeq: 55676 ACK
User-Agent: Grandstream BT100 1.0.5.23
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


13 headers, 0 lines
asterisk1*CLI>

Sip read:
BYE sip:9009 at 208.41.254.125 SIP/2.0
Via: SIP/2.0/UDP 208.41.254.119:5060;
branch=z9hG4bKAClkQjtHCQAPCAAAAAAAAE8A84JLVdR0JbtRLaIFaJU_
Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bK3ed1cbf4ec9bf5be
From: "Shelcomm call forwarding test"
<sip:6262769011 at 208.41.254.119;user=phone>;tag=100c9f35ec6f09a2
To: <sip:9009 at 208.41.254.119;user=phone>;tag=as59b09f62
Proxy-Authorization: DIGEST username="6262769011 at sip.shelcomm.com",
realm="sip.shelcomm.com", algorithm=MD5, uri="sip:9009 at 208.41.254.125",
qop=auth, nc=00000003, cnonce="2eba1ead78e45bc7",
nonce="62JsQimn5xTAMoKDHL+EbAkTRGg=",
response="f18590a3814a4a47757059fe82b75377"
Call-ID: fe67fc663fa36479 at 192.168.1.124
CSeq: 55677 BYE
User-Agent: Grandstream BT100 1.0.5.23
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


12 headers, 0 lines
Sending to 208.41.254.119 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.41.254.119:5060;
branch=z9hG4bKAClkQjtHCQAPCAAAAAAAAE8A84JLVdR0JbtRLaIFaJU_
Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bK3ed1cbf4ec9bf5be
From: "Shelcomm call forwarding test"
<sip:6262769011 at 208.41.254.119;user=phone>;tag=100c9f35ec6f09a2
To: <sip:9009 at 208.41.254.119;user=phone>;tag=as59b09f62
Call-ID: fe67fc663fa36479 at 192.168.1.124
CSeq: 55677 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9009 at 208.41.254.125>
Content-Length: 0


to 208.41.254.119:5060
  == Spawn extension (from-sip-external, 9009, 1) exited non-zero on
'SIP/208.41.254.119-089aef50'
    -- Executing VoiceMail("SIP/208.41.254.119-089aef50", "h") in new stack
    -- Executing Congestion("SIP/208.41.254.119-089aef50", "") in new stack
  == Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/208.41.254.119-089aef50'
Destroying call 'fe67fc663fa36479 at 192.168.1.124'
asterisk1*CLI>

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