[asterisk-dev] Bug 8781 - G729 No path to translate.
Paul Cadach
paul at odt.east.telecom.kz
Sat Feb 10 21:22:37 MST 2007
Did you try to disable G.711 on SIP leg?
WBR,
Paul.
----- Original Message -----
From: "Andy Davidson" <andy at nosignal.org>
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Sent: Saturday, February 10, 2007 3:35 PM
Subject: [asterisk-dev] Bug 8781 - G729 No path to translate.
> 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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