Sip Error 488


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 : 5060 (no NAT)
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port
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 —>
SIP/2.0 488 Not acceptable here[/quote]

Settings sip.conf:

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

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…


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

The only Difference is a other Server IP

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.