[asterisk-dev] Bug 8781 - G729 No path to translate.
Andy Davidson
andy at nosignal.org
Sat Feb 10 16:35:07 MST 2007
Hi,
I think I am suffering from Bug 8781 (grab from the console below.)
I want conversations to flow along this path :
(Wont make sense unless you are using a fixed width font)
[my phone] [asterisk] [third parties]
Snom 360 <----------> v 1.4 <-------------> ???
SIP IAX/SIP
G.729 Don't care (probably something
other than G.729, my preferred
supplier today likes ulaw and
alaw)
My phone sees the * box over a relatively slow consumer connectivity
link. The * box is colocated and has excellent connectivity.
Therefore the tighter compression between * and my phone is
important, hence why I want to use g.729 here. My third party
providers don't support g.729, therefore I have bought a g729
transcode licence. I think once license will be enough as I only
need to transcode once as above.
When I place a call, the other party's line rings as normal. When
the other party answers, I get a sip 'denied' packet, and the call is
aborted. Asterisk says : No path to translate from SIP/mydeskphone
to IAX/myprovider and Had to drop call because I couldn't make SIP/
mydeskphone ompatible with IAX/myprovider. (Full grabs below).
The bug says the try the latest svn version - is this of asterisk
trunk ? I have checked out SVN to revision 53911 and built a new
asterisk.
*CLI> core show version
Asterisk SVN-trunk-r53885 built by root @ billie on a i686 running
Linux on 2007-02-10 23:18:52 UTC
The same thing happens (SIP denied message and accompanied lines on
the console :
[Feb 10 23:34:46] WARNING[19953]: channel.c:3059
ast_channel_make_compatible_helper: No path to translate from SIP/
andydesk-08205dd8(256) to IAX2/213.166.5.130:4569-7(8)
[Feb 10 23:34:46] WARNING[19953]: app_dial.c:1655 dial_exec_full: Had
to drop call because I couldn't make SIP/andydesk-08205dd8 compatible
with IAX2/213.166.5.130:4569-7
)
Any other way I can help debug ?
[ console grab with sip debug ]
billie*CLI>
<--- SIP read from 213.228.240.164:2051 --->
INVITE sip:907738383650 at billie.nosignal.org SIP/2.0
Via: SIP/2.0/UDP 213.228.240.164:2051;branch=z9hG4bK-sm6nj854l6ar;rport
From: "Andy Davidson" <sip:andydesk at billie.nosignal.org>;tag=hx9l95c7l5
To: <sip:907738383650 at billie.nosignal.org>
Call-ID: 3c26c06399cf-k9w8q6ewbb0p at snom360-000413238E15
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:andydesk at 213.228.240.164:2051;line=6wijgpup>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.2.2
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 481
v=0
o=root 1670419164 1670419164 IN IP4 213.228.240.164
s=call
c=IN IP4 213.228.240.164
t=0 0
m=audio 62902 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:L1fDRcmCUTaVHBqf/
EbBLffEF4R3y36XobukKXxz
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv
<------------->
--- (18 headers 19 lines) ---
Sending to 213.228.240.164 : 2051 (NAT)
Using INVITE request as basis request - 3c26c06399cf-
k9w8q6ewbb0p at snom360-000413238E15
<--- Reliably Transmitting (no NAT) to 213.228.240.164:2051 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 213.228.240.164:2051;branch=z9hG4bK-
sm6nj854l6ar;received=213.228.240.164;rport=2051
From: "Andy Davidson" <sip:andydesk at billie.nosignal.org>;tag=hx9l95c7l5
To: <sip:907738383650 at billie.nosignal.org>;tag=as24cb626e
Call-ID: 3c26c06399cf-k9w8q6ewbb0p at snom360-000413238E15
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="0c155afe"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3c26c06399cf-
k9w8q6ewbb0p at snom360-000413238E15' in 32000 ms (Method: INVITE)
Found user 'andydesk'
billie*CLI>
<--- SIP read from 213.228.240.164:2051 --->
ACK sip:907738383650 at billie.nosignal.org SIP/2.0
Via: SIP/2.0/UDP 213.228.240.164:2051;branch=z9hG4bK-sm6nj854l6ar;rport
From: "Andy Davidson" <sip:andydesk at billie.nosignal.org>;tag=hx9l95c7l5
To: <sip:907738383650 at billie.nosignal.org>;tag=as24cb626e
Call-ID: 3c26c06399cf-k9w8q6ewbb0p at snom360-000413238E15
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:andydesk at 213.228.240.164:2051;line=6wijgpup>;flow-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
billie*CLI>
<--- SIP read from 213.228.240.164:2051 --->
INVITE sip:907738383650 at billie.nosignal.org SIP/2.0
Via: SIP/2.0/UDP 213.228.240.164:2051;branch=z9hG4bK-id2bnwj0kolq;rport
From: "Andy Davidson" <sip:andydesk at billie.nosignal.org>;tag=hx9l95c7l5
To: <sip:907738383650 at billie.nosignal.org>
Call-ID: 3c26c06399cf-k9w8q6ewbb0p at snom360-000413238E15
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:andydesk at 213.228.240.164:2051;line=6wijgpup>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.2.2
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest
username="andydesk",realm="asterisk",nonce="0c155afe",uri="sip:
907738383650 at billie.nosignal.org",response="815357e9f93b98a3fa3cdad2506e
b32f",algorithm=md5
Content-Type: application/sdp
Content-Length: 481
v=0
o=root 1670419164 1670419164 IN IP4 213.228.240.164
s=call
c=IN IP4 213.228.240.164
t=0 0
m=audio 62902 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:L1fDRcmCUTaVHBqf/
EbBLffEF4R3y36XobukKXxz
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv
<------------->
--- (19 headers 19 lines) ---
Sending to 213.228.240.164 : 2051 (NAT)
Using INVITE request as basis request - 3c26c06399cf-
k9w8q6ewbb0p at snom360-000413238E15
Found user 'andydesk'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 213.228.240.164:62902
Found description format pcmu for ID 0
Found description format pcma for ID 8
Found description format g722 for ID 9
Found description format g726-32 for ID 2
Found description format gsm for ID 3
Found description format g729 for ID 18
Found description format g723 for ID 4
Found description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x190f
(g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined -
0x10e (gsm|ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.228.240.164:62902
Looking for 907738383650 in default (domain billie.nosignal.org)
list_route: hop: <sip:andydesk at 213.228.240.164:2051;line=6wijgpup>
<--- Transmitting (no NAT) to 213.228.240.164:2051 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.228.240.164:2051;branch=z9hG4bK-
id2bnwj0kolq;received=213.228.240.164;rport=2051
From: "Andy Davidson" <sip:andydesk at billie.nosignal.org>;tag=hx9l95c7l5
To: <sip:907738383650 at billie.nosignal.org>
Call-ID: 3c26c06399cf-k9w8q6ewbb0p at snom360-000413238E15
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:907738383650 at 194.9.77.198>
Content-Length: 0
<------------>
-- Executing [907738383650 at default:1] Set("SIP/
andydesk-082b3728", "CALLERID(num)=+448447047047") in new stack
-- Executing [907738383650 at default:2] Dial("SIP/
andydesk-082b3728", "IAX2/username:password at iax3.magrathea-
telecom.co.uk/07738383650") in new stack
-- Called username:password at iax3.magrathea-telecom.co.uk/
07738383650
-- Call accepted by 213.166.5.130 (format alaw)
-- Format for call is alaw
-- IAX2/213.166.5.130:4569-1 is making progress passing it to
SIP/andydesk-082b3728
Audio is at 194.9.77.198 port 17436
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 213.228.240.164:2051 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 213.228.240.164:2051;branch=z9hG4bK-
id2bnwj0kolq;received=213.228.240.164;rport=2051
From: "Andy Davidson" <sip:andydesk at billie.nosignal.org>;tag=hx9l95c7l5
To: <sip:907738383650 at billie.nosignal.org>;tag=as15eb6895
Call-ID: 3c26c06399cf-k9w8q6ewbb0p at snom360-000413238E15
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:907738383650 at 194.9.77.198>
Content-Type: application/sdp
Content-Length: 332
v=0
o=root 6876 6876 IN IP4 194.9.77.198
s=session
c=IN IP4 194.9.77.198
t=0 0
m=audio 17436 RTP/AVP 18 8 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- IAX2/213.166.5.130:4569-1 answered SIP/andydesk-082b3728
[Feb 10 23:09:27] WARNING[6876]: channel.c:3033
ast_channel_make_compatible: No path to translate from SIP/
andydesk-082b3728(256) to IAX2/213.166.5.130:4569-1(8)
[Feb 10 23:09:27] WARNING[6876]: app_dial.c:1592 dial_exec_full: Had
to drop call because I couldn't make SIP/andydesk-082b3728 compatible
with IAX2/213.166.5.130:4569-1
-- Hungup 'IAX2/213.166.5.130:4569-1'
== Spawn extension (default, 907738383650, 2) exited non-zero on
'SIP/andydesk-082b3728'
Scheduling destruction of SIP dialog '3c26c06399cf-
k9w8q6ewbb0p at snom360-000413238E15' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 213.228.240.164:2051 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 213.228.240.164:2051;branch=z9hG4bK-
id2bnwj0kolq;received=213.228.240.164;rport=2051
From: "Andy Davidson" <sip:andydesk at billie.nosignal.org>;tag=hx9l95c7l5
To: <sip:907738383650 at billie.nosignal.org>;tag=as15eb6895
Call-ID: 3c26c06399cf-k9w8q6ewbb0p at snom360-000413238E15
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:907738383650 at 194.9.77.198>
Content-Length: 0
<------------>
billie*CLI>
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