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Jerry Geis

2008-Apr-01 17:24 UTC

[asterisk-users] help with no audio

I am using asterisk 1.4.18 with a polycom phone.

sip.conf has:
[532]
type=friend
username=532
secret=XXX
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no

I call into the dialplan and try to play demo-congrats and I hear nothing.

Firewall is disabled. 
Everything is on the 192.168.1.X network for this simple configuration.
The tftp server is giving the polycom phone the config files.

Any ideas why I dont hear audio?

Jerry

-----------------------------------

Use 'exit' when done

Asterisk 1.4.18, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
========================================================================  ==
Parsing '/etc/asterisk/asterisk.conf': Found
[0;37;40m[1;30;40m  == [0;37;40mParsing '/etc/asterisk/extconfig.conf':
Found
Connected to Asterisk 1.4.18 currently running on demobox (pid = 18129)
demobox*CLI> 
Verbosity is at least 5

[Kdemobox*CLI> 
    -- Executing [10 at smvoice-sip:1] Playback("SIP/522-051fc8f0",
"demo-congrats") in new stack
?    -- <SIP/522-051fc8f0> Playing 'demo-congrats' (language
'en')
?
[Kdemobox*CLI> 
  == Spawn extension (smvoice-sip, 10, 1) exited non-zero on
'SIP/522-051fc8f0'
?
[Kdemobox*CLI> sip set debug 
demobox*CLI> 
SIP Debugging enabled

[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:10 at 192.168.1.150;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>
CSeq: 1 INVITE
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
?--- (14 headers 11 lines) ---
?
[Kdemobox*CLI> 
Sending to 192.168.1.99 : 5060 (no NAT)
?Using INVITE request as basis request - e6055f35-926fca76-78e0dcbf at
192.168.1.99
?
[Kdemobox*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bK725666cbFE41ACBC;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as47ea6357
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5ac2b96f"
Content-Length: 0


<------------>
?
[Kdemobox*CLI> 
Scheduling destruction of SIP dialog 'e6055f35-926fca76-78e0dcbf at
192.168.1.99' in 32000 ms (Method: INVITE)
?Found user '522'
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
ACK sip:10 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as47ea6357
CSeq: 1 ACK
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (11 headers 0 lines) ---
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:10 at 192.168.1.150;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>
CSeq: 2 INVITE
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="522",
realm="asterisk", nonce="5ac2b96f", uri="sip:10 at
192.168.1.150;user=phone",
response="9a7bd42e9bbf18fef41b63bccc83178c", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
?--- (15 headers 11 lines) ---
?
[Kdemobox*CLI> 
Sending to 192.168.1.99 : 5060 (no NAT)
?Using INVITE request as basis request - e6055f35-926fca76-78e0dcbf at
192.168.1.99
?
[Kdemobox*CLI> 
Found user '522'
?Found RTP audio format 0
?Found RTP audio format 8
?Found RTP audio format 18
?Found RTP audio format 101
?Peer audio RTP is at port 192.168.1.99:2228
?Found audio description format PCMU for ID 0
?Found audio description format PCMA for ID 8
?Found audio description format G729 for ID 18
?Found audio description format telephone-event for ID 101
?Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
?Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
?Peer audio RTP is at port 192.168.1.99:2228
?Looking for 10 in smvoice-sip (domain 192.168.1.150)
?list_route: hop: <sip:522 at 192.168.1.99>
?
<--- Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:10 at 192.168.1.150>
Content-Length: 0


<------------>
?
[Kdemobox*CLI> 
Use 'exit' when done

Asterisk 1.4.18, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
========================================================================  ==
Parsing '/etc/asterisk/asterisk.conf': Found
[0;37;40m[1;30;40m  == [0;37;40mParsing '/etc/asterisk/extconfig.conf':
Found
Connected to Asterisk 1.4.18 currently running on demobox (pid = 18129)
demobox*CLI> 
Verbosity is at least 5

[Kdemobox*CLI> 
    -- Executing [10 at smvoice-sip:1] Playback("SIP/522-051fc8f0",
"demo-congrats") in new stack
?    -- <SIP/522-051fc8f0> Playing 'demo-congrats' (language
'en')
?
[Kdemobox*CLI> 
  == Spawn extension (smvoice-sip, 10, 1) exited non-zero on
'SIP/522-051fc8f0'
?
[Kdemobox*CLI> sip set debug 
demobox*CLI> 
SIP Debugging enabled

[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:10 at 192.168.1.150;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>
CSeq: 1 INVITE
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
?--- (14 headers 11 lines) ---
?
[Kdemobox*CLI> 
Sending to 192.168.1.99 : 5060 (no NAT)
?Using INVITE request as basis request - e6055f35-926fca76-78e0dcbf at
192.168.1.99
?
[Kdemobox*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bK725666cbFE41ACBC;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as47ea6357
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5ac2b96f"
Content-Length: 0


<------------>
?
[Kdemobox*CLI> 
Scheduling destruction of SIP dialog 'e6055f35-926fca76-78e0dcbf at
192.168.1.99' in 32000 ms (Method: INVITE)
?Found user '522'
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
ACK sip:10 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as47ea6357
CSeq: 1 ACK
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (11 headers 0 lines) ---
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:10 at 192.168.1.150;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>
CSeq: 2 INVITE
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="522",
realm="asterisk", nonce="5ac2b96f", uri="sip:10 at
192.168.1.150;user=phone",
response="9a7bd42e9bbf18fef41b63bccc83178c", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
?--- (15 headers 11 lines) ---
?
[Kdemobox*CLI> 
Sending to 192.168.1.99 : 5060 (no NAT)
?Using INVITE request as basis request - e6055f35-926fca76-78e0dcbf at
192.168.1.99
?
[Kdemobox*CLI> 
Found user '522'
?Found RTP audio format 0
?Found RTP audio format 8
?Found RTP audio format 18
?Found RTP audio format 101
?Peer audio RTP is at port 192.168.1.99:2228
?Found audio description format PCMU for ID 0
?Found audio description format PCMA for ID 8
?Found audio description format G729 for ID 18
?Found audio description format telephone-event for ID 101
?Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
?Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
?Peer audio RTP is at port 192.168.1.99:2228
?Looking for 10 in smvoice-sip (domain 192.168.1.150)
?list_route: hop: <sip:522 at 192.168.1.99>
?
<--- Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:10 at 192.168.1.150>
Content-Length: 0


<------------>
?
[Kdemobox*CLI> 
    -- Executing [10 at smvoice-sip:1] Playback("SIP/522-051fc8f0",
"demo-congrats") in new stack
?
[Kdemobox*CLI> 
Audio is at 192.168.1.150 port 16160
?
[Kdemobox*CLI> 
Adding codec 0x4 (ulaw) to SDP
?
[Kdemobox*CLI> 
Adding codec 0x8 (alaw) to SDP
?
[Kdemobox*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
?
[Kdemobox*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:10 at 192.168.1.150>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 18129 18129 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 16160 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 - - - -
a=ptime:20
a=sendrecv

<------------>
?
[Kdemobox*CLI> 
    -- <SIP/522-051fc8f0> Playing 'demo-congrats' (language
'en')
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
ACK sip:10 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bKc2d2ee271CDE1AF8
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
CSeq: 2 ACK
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Proxy-Authorization: Digest username="522",
realm="asterisk", nonce="5ac2b96f", uri="sip:10 at
192.168.1.150;user=phone",
response="9a7bd42e9bbf18fef41b63bccc83178c", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (12 headers 0 lines) ---
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
BYE sip:10 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK495a4b51BC858372
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
CSeq: 3 BYE
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Proxy-Authorization: Digest username="522",
realm="asterisk", nonce="5ac2b96f", uri="sip:10 at
192.168.1.150;user=phone",
response="7c294ada2c2faf290989d697a6b26282", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (11 headers 0 lines) ---
?
[Kdemobox*CLI> 
Sending to 192.168.1.99 : 5060 (no NAT)
?
[Kdemobox*CLI> 
<--- Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bK495a4b51BC858372;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:10 at 192.168.1.150>
Content-Length: 0


<------------>
?
[Kdemobox*CLI> 
  == Spawn extension (smvoice-sip, 10, 1) exited non-zero on
'SIP/522-051fc8f0'
?
[Kdemobox*CLI> 
Really destroying SIP dialog 'e6055f35-926fca76-78e0dcbf at
192.168.1.99' Method: BYE
?
[Kdemobox*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
    -- Executing [10 at smvoice-sip:1] Playback("SIP/522-051fc8f0",
"demo-congrats") in new stack
?
[Kdemobox*CLI> 
Audio is at 192.168.1.150 port 16160
?
[Kdemobox*CLI> 
Adding codec 0x4 (ulaw) to SDP
?
[Kdemobox*CLI> 
Adding codec 0x8 (alaw) to SDP
?
[Kdemobox*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
?
[Kdemobox*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:10 at 192.168.1.150>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 18129 18129 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 16160 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 - - - -
a=ptime:20
a=sendrecv

<------------>
?
[Kdemobox*CLI> 
    -- <SIP/522-051fc8f0> Playing 'demo-congrats' (language
'en')
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
ACK sip:10 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bKc2d2ee271CDE1AF8
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
CSeq: 2 ACK
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Proxy-Authorization: Digest username="522",
realm="asterisk", nonce="5ac2b96f", uri="sip:10 at
192.168.1.150;user=phone",
response="9a7bd42e9bbf18fef41b63bccc83178c", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (12 headers 0 lines) ---
?
[Kdemobox*CLI> 
<--- SIP read from 192.168.1.99:5060 --->
BYE sip:10 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK495a4b51BC858372
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
CSeq: 3 BYE
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
Contact: <sip:522 at 192.168.1.99>
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Proxy-Authorization: Digest username="522",
realm="asterisk", nonce="5ac2b96f", uri="sip:10 at
192.168.1.150;user=phone",
response="7c294ada2c2faf290989d697a6b26282", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (11 headers 0 lines) ---
?
[Kdemobox*CLI> 
Sending to 192.168.1.99 : 5060 (no NAT)
?
[Kdemobox*CLI> 
<--- Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bK495a4b51BC858372;received=192.168.1.99
From: "522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969
To: <sip:10 at 192.168.1.150;user=phone>;tag=as6dc593ac
Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:10 at 192.168.1.150>
Content-Length: 0


<------------>
?
[Kdemobox*CLI> 
  == Spawn extension (smvoice-sip, 10, 1) exited non-zero on
'SIP/522-051fc8f0'
?
[Kdemobox*CLI> 
Really destroying SIP dialog 'e6055f35-926fca76-78e0dcbf at
192.168.1.99' Method: BYE
?
[Kdemobox*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).

Jared Smith

2008-Apr-01 17:32 UTC

head link

[asterisk-users] help with no audio

On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:> I call into the dialplan and try to play demo-congrats and I hear nothing.
> 
> Firewall is disabled. 
> Everything is on the 192.168.1.X network for this simple configuration.
> The tftp server is giving the polycom phone the config files.
> 
> Any ideas why I dont hear audio?
Do you happen to have an unconfigured T1 card in your machine?  That's
the most common problem I see for people when they get no audio at all
coming out of Asterisk.  If that's not the case, I'd turn RTP debugging
on in the command-line and make sure RTP packets are coming and going
from the Asterisk box.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.

Jerry Geis

2008-Apr-01 17:44 UTC

head link

[asterisk-users] help with no audio

>
> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
> >/ I call into the dialplan and try to play demo-congrats and I hear
nothing.
> />/ 
> />/ Firewall is disabled. 
> />/ Everything is on the 192.168.1.X network for this simple
configuration.
> />/ The tftp server is giving the polycom phone the config files.
> />/ 
> />/ Any ideas why I dont hear audio?
> /
> Do you happen to have an unconfigured T1 card in your machine?  That's
> the most common problem I see for people when they get no audio at all
> coming out of Asterisk.  If that's not the case, I'd turn RTP
debugging
> on in the command-line and make sure RTP packets are coming and going
> from the Asterisk box.
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
Jared,

I have no card in this unit at this time.
lsmod shows ztdummy loaded.

I turned on "rtp debug" and got a bunch of lines like:
Got RTP packet from 192.168.1.99:2226 ....

Does that help you?

Jerry

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