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I have installed an asterisk it repeats this statement, again and again, my server is overloaded because of these errors.

[Sep 19 18:34:23] NOTICE[1641][C-00000398]: chan_sip.c:10995 process_sdp: No compatible codecs, not accepting this offer!

Some one please help me.

--- (10 headers 0 lines) ---
Really destroying SIP dialog '0dd03af063616c6c00a3c90e@94.130.128.46' Method: ACK

<--- SIP read from UDP:111.111.128.46:5060 --->
INVITE sip:447536913595@80.209.237.173 SIP/2.0
Via: SIP/2.0/UDP 111.111.128.46:5060;branch=z9hG4bK5814af823fb470c7
From: <sip:442075304284@111.111.128.46>;tag=77b347fa5b331553
To: <sip:447536913595@80.209.237.173>
Call-ID: 03a42c5d63616c6c00a3c912@111.111.128.46
CSeq: 21703 INVITE
Contact: <sip:442075304284@111.111.128.46:5060>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer, linknat
Original-Info: hIEtgm6x4Rq4wFcwf0ECXFlXQQceAk8GX+Ku
Max-Forwards: 70
User-Agent: VOS3000 V2.1.7.03
Session-Expires: 600
Original-ID: 17b43a2f-11b45755-7b663-7fc95ce87268-8a534841-13c4-7225
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 36535 36535 IN IP4 111.111.128.46
s=VOS3000
c=IN IP4 111.111.128.46
t=0 0
m=audio 37366 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (16 headers 11 lines) ---
Sending to 111.111.128.46:5060 (no NAT)
Sending to 111.111.128.46:5060 (no NAT)
Using INVITE request as basis request - 03a42c5d63616c6c00a3c912@94.130.128.46
No matching peer for '442075304284' from '111.111.128.46:5060'
  == Using SIP RTP CoS mark 5
Got SDP version 36535 and unique parts [- 36535 IN IP4 94.130.128.46]
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
[Sep 19 18:45:59] NOTICE[1641][C-0000114a]: chan_sip.c:10995 process_sdp: No compatible codecs, not accepting this offer!

<--- Reliably Transmitting (no NAT) to 111.111.128.46:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 94.130.128.46:5060;branch=z9hG4bK5814af823fb470c7;received=94.130.128.46
From: <sip:442075304284@111.111.128.46>;tag=77b347fa5b331553
To: <sip:447536913595@80.209.237.173>;tag=as7a56c326
Call-ID: 03a42c5d63616c6c00a3c912@94.130.128.46
CSeq: 21703 INVITE
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0```
Huzoor Bux
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1 Answers1

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Your device (VOS3000 V2.1.7.03) at 111.111.128.46 is configured to only offer a single voice codec, G729:

m=audio 37366 RTP/AVP 18 101
a=rtpmap:18 G729/8000

However I suspect you Asterisk box has no G729 module and/or licence. You can check:

*CLI> core show translation paths g729
--- Translation paths SRC Codec "g729" sample rate 8000 ---

Also make sure the SIP peer has allow=g729 configured as well.

But I would recommend to configure that device offer G711, which is more common.

viktike
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