Sip Error 488

Isip · May 13, 2009, 5:53am

Hello,

I have problems with dids on the asterisk, G 729 is supported
(and white any other did frome the same provider and G 729 running this did whitout trouble)

sip debug log:

[quote]Sending to 190.6.xxx.xxx : 5060 (no NAT)
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port 190.6.xxx.xxx:16586
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 19
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)

<— Reliably Transmitting (no NAT) to 190.6.xxx.xxx:5060 —>
SIP/2.0 488 Not acceptable here[/quote]

Settings sip.conf:
[/quote]
[test1]
type=peer
host=190.6.xxx.xxx
context=test1
disallow=all
allow=ulaw
allow=g729
[/quote]

who can help me ?[ul][/ul]

g2010 · May 13, 2009, 12:33pm

The core problem here is this line:

Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing)

Your asterisk box is allow gsm,ulaw,alaw,h263
Your peer is only allow g729

The SIP settings you posted do NOT reflect the SIP messaging for this peer, which is why there is probably a peer/trunk specific section which is overriding what you posted. Please note, that if you have two peers defined with the same IP address, asterisk will match against the last one in the configuration file…

Isip · May 13, 2009, 2:46pm

hello,

the sip-debug from the failed did are not complete,because
not more message from the debug

but with any other did from the same provider and the same sip.config settings this did running without trouble:

Settings sip.conf:

and the debug from this did-test

<------------->
— (13 headers 11 lines) —
Sending to 190.6.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request - 273159d417856255238153966048a9a4@190.xxx.xxx.xxx
Found peer ‘test’
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 190.6.xxx.xxx:17192
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 190.6.xxx.xxx:17192
Looking for 5726205555 in didww (domain 82.xxx.xxx.xxx)
list_route: hop: sip:01157300654433@190.6.xxx.xxx

<— Transmitting (no NAT) to 190.6.xxx.xxx:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.6.xxx.xxx:5060;branch=z9hG4bK031378e0;received=190.6.xxx.xxx;rport=5060
From: “01157300654433” sip:01157300654433@190.6.xxx.xxx;tag=as306a93da
To: sip:5726205555@82.xxx.xxx.xxx
Call-ID: 273159d417856255238153966048a9a4@190.6.xxx.xxx
CSeq: 102 INVITE
User-Agent: user
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:5726205555@82.xxx.xxx.xxx
Content-Length: 0

<------------>
– Executing [5726205555@didww:1] NoOp(“SIP/190.6.xxx.xxx-081f8510”, “01157300654433”) in new stack
– Executing [5726205555@didww:2] Dial(“SIP/190.6.xxx.xxx-081f8510”, “SIP/2000”) in new stack
Audio is at 82.xxx.xxx.xxx port 13204
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 190.xxx.xxx.xxx:5060:
INVITE sip:2000@190.xxx.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.xxx.xxx.xxx:5060;branch=z9hG4bK276fe21e;rport
From: “01157300654433” sip:01157300654433@82.xxx.xxx.xxx;tag=as680fc859
To: sip:2000@190.xxx.xxx.xxx:5060;transport=udp
Contact: sip:01157300654433@82.xxx.xxx.xxx
Call-ID: 212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: user
Max-Forwards: 70
Date: Wed, 13 May 2009 14:13:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 13132 13132 IN IP4 82.xxx.xxx.xxx
s=session
c=IN IP4 82.xxx.xxx.xxx
t=0 0
m=audio 13204 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
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
-- Called 2000
v90460*CLI>
<— SIP read from 190.xxx.xxx.xxx:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.xxx.xxx.xxx:5060;branch=z9hG4bK276fe21e;rport
From: “01157300654433” sip:01157300654433@82.xxx.xxx.xxx;tag=as680fc859
To: sip:2000@190.xxx.xxx.xxx:5060;transport=udp
Call-ID: 212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.6.32
Content-Length: 0

<------------->
— (8 headers 0 lines) —
v90460*CLI>
<— SIP read from 190.xxx.xxx.xxx:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 82.xxx.xxx.xxx:5060;branch=z9hG4bK276fe21e;rport
From: “01157300654433” sip:01157300654433@82.xxx.xxx.xxx;tag=as680fc859
To: sip:2000@190.xxx.xxx.xxx:5060;transport=udp;tag=1f40945017631073
Call-ID: 212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.6.32
Contact: sip:2000@190.xxx.xxx.xxx:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– SIP/2000-081fc518 is ringing

<— Transmitting (no NAT) to 190.6.xxx.xxx:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 190.6.xxx.xxx:5060;branch=z9hG4bK031378e0;received=190.6.xxx.xxx;rport=5060
From: “011573005621781” sip:01157300654433@190.6.xxx.xxx;tag=as306a93da
To: sip:5726205555@82.xxx.xxx.xxx;tag=as3ae23a61
Call-ID: 273159d417856255238153966048a9a4@190.6.xxx.xxx
CSeq: 102 INVITE
User-Agent: user
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:5726205555@82.xxx.xxx.xxx
Content-Length: 0

<------------>
v90460*CLI>
<— SIP read from 190.6.xxx.xxx:5060 —>
CANCEL sip:5726205555@82.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 190.6.xxx.xxx:5060;branch=z9hG4bK031378e0;rport
From: “011573005621781” sip:01157300654433@190.6.xxx.xxx;tag=as306a93da
To: sip:5726205555@82.xxx.xxx.xxx
Call-ID: 273159d417856255238153966048a9a4@190.6.xxx.xxx
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 190.6.xxx.xxx : 5060 (NAT)

<— Reliably Transmitting (NAT) to 190.6.xxx.xxx:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 190.6.xxx.xxx.242:5060;branch=z9hG4bK031378e0;received=190.6.xxx.xxx;rport=5060
From: “011573005621781” sip:01157300654433@190.6.xxx.xxx;tag=as306a93da
To: sip:5726205555@82.xxx.xxx.xxx;tag=as3ae23a61
Call-ID: 273159d417856255238153966048a9a4@190.6.xxx.xxx
CSeq: 102 INVITE
User-Agent: user
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>

<— Transmitting (NAT) to 190.6.xxx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.6.xxx.xxx:5060;branch=z9hG4bK031378e0;received=190.6.xxx.xxx;rport=5060
From: “01157300654433” sip:01157300654433@190.6.xxx.xxx;tag=as306a93da
To: sip:5726205555@82.xxx.xxx.xxx;tag=as3ae23a61
Call-ID: 273159d417856255238153966048a9a4@190.6.xxx.xxx
CSeq: 102 CANCEL
User-Agent: user
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx’ in 15872 ms (Method: INVITE)
Reliably Transmitting (NAT) to 190.xxx.xxx.xxx:5060:
CANCEL sip:2000@190.xxx.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.xxx.xxx.xxx:5060;branch=z9hG4bK276fe21e;rport
From: “01157300654433” sip:01157300654433@82.xxx.xxx.xxx;tag=as680fc859
To: sip:2000@190.xxx.xxx.xxx:5060;transport=udp
Call-ID: 212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx
CSeq: 102 CANCEL
User-Agent: user
Max-Forwards: 70
Content-Length: 0

Scheduling destruction of SIP dialog '212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx’ in 15872 ms (Method: INVITE)
== Spawn extension (didww, 5726205555, 2) exited non-zero on ‘SIP/190.6.160.242-081f8510’
v90460*CLI>
<— SIP read from 190.6.xxx.xxx:5060 —>
ACK sip:5726205555@82.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 190.xxx.xxx.xxx:5060;branch=z9hG4bK031378e0;rport
From: “01157300654433” sip:01157300654433@190.6.xxx.xxx;tag=as306a93da
To: sip:5726205555@82.xxx.xxx.xxx;tag=as3ae23a61
Contact: sip:011573005621781@190.6.xxx.xxx
Call-ID: 273159d417856255238153966048a9a4@190.6.xxx.xxx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog '273159d417856255238153966048a9a4@190.6.xxx.xxx’ Method: ACK
v90460*CLI>
<— SIP read from 190.xxx.xxx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.xxx.xxx.xxx:5060;branch=z9hG4bK276fe21e;rport
From: “01157300654433” sip:01157300654433@82.xxx.xxx.xxx;tag=as680fc859
To: sip:2000@190.xxx.xxx.xxx:5060;transport=udp;tag=1f40945017631073
Call-ID: 212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx
CSeq: 102 CANCEL
User-Agent: Grandstream BT200 1.1.6.32
Supported: replaces, timer
Content-Length: 0

<------------->
— (9 headers 0 lines) —
v90460*CLI>
<— SIP read from 190.6.xxx.xxx:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 82.xxx.xxx.xxx:5060;branch=z9hG4bK276fe21e;rport
From: “01157300654433” sip:01157300654433@82.xxx.xxx.xxx;tag=as680fc859
To: sip:2000@190.xxx.xxx.xxx:5060;transport=udp;tag=1f40945017631073
Call-ID: 212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.6.32
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 190.xxx.xxx.xxx:5060:
ACK sip:2000@190.xxx.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.xxx.xxx.xxx:5060;branch=z9hG4bK276fe21e;rport
From: “01157300654433” sip:01157300654433@82.xxx.xxx.xxx;tag=as680fc859
To: sip:2000@190.xxx.xxx.xxx:5060;transport=udp;tag=1f40945017631073
Contact: sip:01157300654433@82.xxx.xxx.xxx
Call-ID: 212aef84233fb636507af4d27b79275d@82.xxx.xxx.xxx
CSeq: 102 ACK
User-Agent: user
Max-Forwards: 70
Content-Length: 0

The only Difference is a other Server IP

g2010 · May 13, 2009, 2:50pm

Post up your entire sip.conf and sip_custom.conf (or something like that if running a GUI)

Also, post the entire SIP transaction including the initial INVITE for both a working and non-working call.

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Sip Error 488

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