[Asterisk-Users] Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO
Steve Ducat
steveducat at gmail.com
Thu Sep 29 13:30:17 MST 2005
OK, here goes my next problem.
I have puchased a DID which I can connect to via SIP
I have been given the following details:
Username: uka1xxxxxx
Password: 1000xxxxxx
Server: brxxxx.net:5160
My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)
The other end is a Cisco AS5300 (NO NAT)
I can register with the Cisco with no problem.
When I dial the DID it sends the call to my asterisk server and my
asterisk server sends back the dial tone, no problem.
The problem is when I pick up the phone, no audio.
If I change the dial plan to do a Playback instead of Dial an
extension I can see in the console it answers the call and starts to
play the Playback but no audio.
I can connect direclty to the Cisco AS5300 with sjphone or a budgetone
102 with no problem and get dial tone and full audio both ways but
when I use the asterisk no audio.
I have tried every codec possible. I have installed g729, g723 with no
luck. I have tested both these codecs by forcing my budgetone to use
with no problem so I know the codecs work.
So the problem is when I ask asterisk to register to the Cisco AS5300
as a SIP Client it does everything right except pass the audio.
There is no firewall configured.
I know the Cisco SIP Server works because it works with the softphone
SJPHONE and directly with the Budgetone 102.
I have reinstalled Asterisk so many times.
I have reinstalled g729 & g723 so many times.
The SIP debug output is pasted below.
I have been struggling with this for weeks with no luck.
Any help would be appreciated.
Steven Ducat.
*********************************************************************
<-- SIP read from 203.88.192.42:5160:
INVITE sip:84104214 at 70.84.200.204 SIP/2.0
Record-Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786
From: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214 at 203.88.192.42>
Date: Thu, 29 Sep 2005 20:14:40 GMT
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 2153363387-811340250-2169109749-53752559
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 5
Remote-Party-ID:
<sip:0017911 at 211.147.240.237>;party=calling;screen=yes;privacy=off
Timestamp: 1128024880
Contact: <sip:0017911 at 211.147.240.237:57786>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 432
P-hint: Proxied
P-hint: usrloc applied
v=0
o=CiscoSystemsSIP-GW-UserAgent 5786 3481 IN IP4 211.147.240.237
s=SIP Call
c=IN IP4 211.147.240.237
t=0 0
m=audio 37708 RTP/AVP 18 4 3 8 0 110
c=IN IP4 203.88.192.42
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 X-NSE/8000
a=fmtp:110 192-194
a=direction:passive
a=direction:active
a=nortpproxy:yes
--- (24 headers 19 lines)---
Using INVITE request as basis request -
805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
Sending to 203.88.192.42 : 5160 (non-NAT)
Found no matching peer or user for '203.88.192.42:5160'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 110
Peer audio RTP is at port 211.147.240.237:37708
Found description format G729
Found description format G723
Found description format GSM
Found description format PCMA
Found description format PCMU
Found description format X-NSE
Capabilities: us - 0x100 (g729), peer - audio=0x30f
(g723|gsm|ulaw|alaw|g729|speex)/video=0x0 (nothing), combined - 0x100
(g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Looking for 84104214 in default (domain 70.84.200.204)
list_route: hop: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
Transmitting (no NAT) to 203.88.192.42:5160:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786
From: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214 at 203.88.192.42>
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:84104214 at 70.84.200.204>
Content-Length: 0
---
-- Executing Dial("SIP/211.147.240.237-b7116c10", "Local/2001/n")
in new stack
-- Executing Macro("Local/2001 at default-acc1,2",
"oneline|SIP/stevenducat") in new stack
-- Executing Dial("Local/2001 at default-acc1,2",
"SIP/stevenducat|20") in new stack
-- Called 2001/n
We're at 70.84.200.204 port 14922
Answering/Requesting with root capability 0x100 (g729)
12 headers, 8 lines
Reliably Transmitting (NAT) to 83.146.11.93:60073:
INVITE sip:stevenducat at 192.168.0.7:18234 SIP/2.0
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport
From: "0017911" <sip:stevenducat at 70.84.200.204>;tag=as2c8caf36
To: <sip:stevenducat at 192.168.0.7:18234>
Contact: <sip:stevenducat at 70.84.200.204>
Call-ID: 438558184cf076d15a92ff5831e2d285 at 70.84.200.204
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Sep 2005 20:18:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 164
v=0
o=root 14260 14260 IN IP4 70.84.200.204
s=session
c=IN IP4 70.84.200.204
t=0 0
m=audio 14922 RTP/AVP 18
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
---
-- Called stevenducat
usa*CLI>
<-- SIP read from 83.146.11.93:60073:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport
From: "0017911" <sip:stevenducat at 70.84.200.204>;tag=as2c8caf36
To: <sip:stevenducat at 192.168.0.7:18234>
Call-ID: 438558184cf076d15a92ff5831e2d285 at 70.84.200.204
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Content-Length: 0
--- (8 headers 0 lines)---
usa*CLI>
<-- SIP read from 83.146.11.93:60073:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport
From: "0017911" <sip:stevenducat at 70.84.200.204>;tag=as2c8caf36
To: <sip:stevenducat at 192.168.0.7:18234>;tag=069c9468d5984dbc
Call-ID: 438558184cf076d15a92ff5831e2d285 at 70.84.200.204
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Content-Length: 0
--- (8 headers 0 lines)---
-- SIP/stevenducat-26a1 is ringing
-- Local/2001 at default-acc1,1 is ringing
Transmitting (no NAT) to 203.88.192.42:5160:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786
From: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:84104214 at 70.84.200.204>
Content-Length: 0
---
usa*CLI>
<-- SIP read from 203.88.192.42:5160:
--- (0 headers 0 lines) Nat keepalive ---
usa*CLI>
<-- SIP read from 203.88.192.42:5160:
--- (0 headers 0 lines) Nat keepalive ---
usa*CLI>
<-- SIP read from 83.146.11.93:60073:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport
From: "0017911" <sip:stevenducat at 70.84.200.204>;tag=as2c8caf36
To: <sip:stevenducat at 192.168.0.7:18234>;tag=069c9468d5984dbc
Call-ID: 438558184cf076d15a92ff5831e2d285 at 70.84.200.204
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Contact: <sip:stevenducat at 192.168.0.7:18234>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 162
v=0
o=stevenducat 8000 8000 IN IP4 192.168.0.7
s=SIP Call
c=IN IP4 192.168.0.7
t=0 0
m=audio 6290 RTP/AVP 18
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:20
--- (12 headers 9 lines)---
Found RTP audio format 18
Peer audio RTP is at port 192.168.0.7:6290
Found description format G729
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0
(nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
list_route: hop: <sip:stevenducat at 192.168.0.7:18234>
set_destination: Parsing <sip:stevenducat at 192.168.0.7:18234> for
address/port to send to
set_destination: set destination to 192.168.0.7, port 18234
Transmitting (NAT) to 83.146.11.93:60073:
ACK sip:stevenducat at 192.168.0.7:18234 SIP/2.0
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK3a80325d;rport
From: "0017911" <sip:stevenducat at 70.84.200.204>;tag=as2c8caf36
To: <sip:stevenducat at 192.168.0.7:18234>;tag=069c9468d5984dbc
Contact: <sip:stevenducat at 70.84.200.204>
Call-ID: 438558184cf076d15a92ff5831e2d285 at 70.84.200.204
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
---
-- SIP/stevenducat-26a1 answered Local/2001 at default-acc1,2
-- Local/2001 at default-acc1,1 stopped sounds
-- Local/2001 at default-acc1,1 answered SIP/211.147.240.237-b7116c10
We're at 70.84.200.204 port 10134
Answering with preferred capability 0x100 (g729)
Reliably Transmitting (no NAT) to 203.88.192.42:5160:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786
Record-Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
From: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:84104214 at 70.84.200.204>
Content-Type: application/sdp
Content-Length: 164
v=0
o=root 14260 14260 IN IP4 70.84.200.204
s=session
c=IN IP4 70.84.200.204
t=0 0
m=audio 10134 RTP/AVP 18
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
---
usa*CLI>
<-- SIP read from 203.88.192.42:5160:
ACK sip:84104214 at 70.84.200.204:5060 SIP/2.0
Via: SIP/2.0/UDP 203.88.192.42:5160;branch=0
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=56631
From: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
Date: Thu, 29 Sep 2005 20:14:40 GMT
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
Max-Forwards: 5
Content-Length: 0
CSeq: 101 ACK
P-hint: Proxied
P-hint: rr-enforced
--- (12 headers 0 lines)---
usa*CLI>
<-- SIP read from 83.146.11.93:60073:
--- (0 headers 0 lines) Nat keepalive ---
Destroying call '3d52632b21629e691095c20515b4847b at 70.84.200.204'
Destroying call '2ed06e446b9b703e63a07a783dc552ea at 70.84.200.204'
usa*CLI>
<-- SIP read from 83.146.11.93:60073:
BYE sip:stevenducat at 70.84.200.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7:18234;branch=z9hG4bK0778411ca2ccfa45
From: <sip:stevenducat at 192.168.0.7:18234>;tag=069c9468d5984dbc
To: "0017911" <sip:stevenducat at 70.84.200.204>;tag=as2c8caf36
Call-ID: 438558184cf076d15a92ff5831e2d285 at 70.84.200.204
CSeq: 32201 BYE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (10 headers 0 lines)---
Sending to 192.168.0.7 : 18234 (NAT)
Transmitting (NAT) to 83.146.11.93:60073:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.7:18234;branch=z9hG4bK0778411ca2ccfa45;received=83.146.11.93
From: <sip:stevenducat at 192.168.0.7:18234>;tag=069c9468d5984dbc
To: "0017911" <sip:stevenducat at 70.84.200.204>;tag=as2c8caf36
Call-ID: 438558184cf076d15a92ff5831e2d285 at 70.84.200.204
CSeq: 32201 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:stevenducat at 70.84.200.204>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
== Spawn extension (macro-oneline, s, 1) exited non-zero on
'Local/2001 at default-acc1,2' in macro 'oneline'
== Spawn extension (default, 2001, 1) exited non-zero on
'Local/2001 at default-acc1,2'
== Spawn extension (default, 84104214, 1) exited non-zero on
'SIP/211.147.240.237-b7116c10'
set_destination: Parsing
<sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on> for address/port to
send to
set_destination: set destination to 203.88.192.42, port 5160
Reliably Transmitting (no NAT) to 203.88.192.42:5160:
BYE sip:0017911 at 211.147.240.237:57786 SIP/2.0
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK1d68194f;rport
Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
From: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
To: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
Contact: <sip:84104214 at 70.84.200.204>
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
---
Destroying call '438558184cf076d15a92ff5831e2d285 at 70.84.200.204'
Retransmitting #1 (no NAT) to 203.88.192.42:5160:
BYE sip:0017911 at 211.147.240.237:57786 SIP/2.0
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK1d68194f;rport
Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
From: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
To: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
Contact: <sip:84104214 at 70.84.200.204>
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
---
usa*CLI>
<-- SIP read from 203.88.192.42:5160:
BYE sip:84104214 at 70.84.200.204:5060 SIP/2.0
Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK2aeb.eb8a2852.0
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=50629
From: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
Date: Thu, 29 Sep 2005 20:14:40 GMT
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 5
Timestamp: 1128024902
CSeq: 102 BYE
Content-Length: 0
P-hint: Proxied
P-hint: rr-enforced
--- (14 headers 0 lines)---
Sending to 203.88.192.42 : 5160 (non-NAT)
Transmitting (no NAT) to 203.88.192.42:5160:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
203.88.192.42:5160;branch=z9hG4bK2aeb.eb8a2852.0;received=203.88.192.42
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=50629
From: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:84104214 at 70.84.200.204>
Content-Length: 0
---
usa*CLI>
<-- SIP read from 203.88.192.42:5160:
--- (0 headers 0 lines) Nat keepalive ---
<-- SIP read from 203.88.192.42:5160:
--- (0 headers 0 lines) Nat keepalive ---
Retransmitting #2 (no NAT) to 203.88.192.42:5160:
BYE sip:0017911 at 211.147.240.237:57786 SIP/2.0
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK1d68194f;rport
Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
From: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
To: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
Contact: <sip:84104214 at 70.84.200.204>
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
---
Retransmitting #3 (no NAT) to 203.88.192.42:5160:
BYE sip:0017911 at 211.147.240.237:57786 SIP/2.0
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK1d68194f;rport
Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
From: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
To: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
Contact: <sip:84104214 at 70.84.200.204>
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
---
usa*CLI>
<-- SIP read from 203.88.192.42:5160:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP
70.84.200.204:5060;branch=z9hG4bK1d68194f;rport=5060;received=70.84.200.202
From: <sip:84104214 at 203.88.192.42>;tag=as2ab4875c
To: <sip:0017911 at 211.147.240.237>;tag=1CA65AC-9C8
Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237
CSeq: 102 BYE
Server: Broadz SIP Proxy (0.10.0-dev12 (i386/linux))
Content-Length: 0
--- (8 headers 0 lines)---
SIP Response message for INCOMING dialog BYE arrived
-- Incoming call: Got SIP response 408 "Request Timeout" back from
203.88.192.42
Destroying call '805AF00B-305C11DA-814CFCF5-33432EF at 211.147.240.237'
usa*CLI> sip no debug
SIP Debugging Disabled
usa*CLI>
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