Sip 488 not acceptable here что это такое
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Sip 488 not acceptable here что это такое

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Sip 488 not acceptable here что это такое

Новичком считается только что прочитавший «Астериск — будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модератор: april22

Сообщений: 6 • Страница 1 из 1

SIP/2.0 488 Not acceptable here — причем только с Мегафона

minidisk » 18 янв 2016, 15:58

Всех приветствую и надеюсь на помощь)))
Проблема такова..
Есть Asterisk 1.8.20 (Elastix 2.4.0), настроен — работает, но при звонке ТОЛЬКО с Мегафона на астериск выскакивает SIP/2.0 488 Not acceptable here.

Codecs : 0x10e (gsm|ulaw|alaw|g729)
Codec Order : (alaw:60,ulaw:20,gsm:60,g729:60)

<--- SIP read from UDP:19.2.41.21:5060 --->
INVITE sip:s@1.2.3.4:5060;received=1.2.3.4:41394 SIP/2.0
Via: SIP/2.0/UDP 19.2.41.21;rport;branch=z9hG4bK0gvvQyg961BjS
Route:
Max-Forwards: 68
From: «+79262719630» ;tag=XaggZpUQ1m3vm
To:
Call-ID: 33783cca-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Contact:
User-Agent: MoLi callswitch1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 373
X-FS-Support: update_display,send_info
Remote-Party-ID: «+79262719630» ;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1453086816 1453086817 IN IP4 19.2.41.21
s=FreeSWITCH
c=IN IP4 19.2.41.21
t=0 0
m=audio 30020 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off — — — —
a=ptime:30
m=audio 30020 RTP/AVP 8 0 18 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off — — — —
a=ptime:20

— (18 headers 16 lines) —
Sending to 19.2.41.21:5060 (NAT)
Using INVITE request as basis request — 33783cca-387a-1234-798b-002590e5052c
Found peer ‘M_010’ for ‘89262719630’ from 19.2.41.21:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 19.2.41.21;branch=z9hG4bK0gvvQyg961BjS;received=19.2.41.21;rport=5060
From: «+79262719630» ;tag=XaggZpUQ1m3vm
To: ;tag=as1471402a
Call-ID: 33783cca-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Scheduling destruction of SIP dialog ‘33783cca-387a-1234-798b-002590e5052c’ in 6400 ms (Method: INVITE)

ACK sip:s@1.2.3.4:5060;received=1.2.3.4:41394 SIP/2.0
Via: SIP/2.0/UDP 19.2.41.21;rport;branch=z9hG4bK0gvvQyg961BjS
Route:
Max-Forwards: 68
From: «+79262719630» ;tag=XaggZpUQ1m3vm
To: ;tag=as1471402a
Call-ID: 33783cca-387a-1234-798b-002590e5052c
CSeq: 86222354 ACK
Content-Length: 0

— (9 headers 0 lines) —
Really destroying SIP dialog ‘33783cca-387a-1234-798b-002590e5052c’ Method: ACK

INVITE sip:s@1.2.3.4:5060;received=1.2.3.4:41394 SIP/2.0
Via: SIP/2.0/UDP 19.2.41.21;rport;branch=z9hG4bK1SNNSS1c4a24m
Route:
Max-Forwards: 68
From: «+79262719630» ;tag=Zv212cXyU6F2B
To:
Call-ID: 337bce93-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Contact:
User-Agent: MoLie callswitch1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 373
X-FS-Support: update_display,send_info
Remote-Party-ID: «+79262719630» ;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1453097848 1453097849 IN IP4 193.27.41.21
s=FreeSWITCH
c=IN IP4 19.2.41.21
t=0 0
m=audio 18988 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off — — — —
a=ptime:30
m=audio 18988 RTP/AVP 8 0 18 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off — — — —
a=ptime:20

— (18 headers 16 lines) —
Sending to 19.2.41.21:5060 (NAT)
Using INVITE request as basis request — 337bce93-387a-1234-798b-002590e5052c
Found peer ‘Mo_010’ for ‘89262719630’ from 19.2.41.21:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 19.2.41.21;branch=z9hG4bK1SNNSS1c4a24m;received=19.2.41.21;rport=5060
From: «+79262719630» ;tag=Zv212cXyU6F2B
To: ;tag=as4f0d2223
Call-ID: 337bce93-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Scheduling destruction of SIP dialog ‘337bce93-387a-1234-798b-002590e5052c’ in 6400 ms (Method: INVITE)

ACK sip:s@1.2.3.4:5060;received=1.2.3.4:41394 SIP/2.0
Via: SIP/2.0/UDP 19.2.41.21;rport;branch=z9hG4bK1SNNSS1c4a24m
Route:
Max-Forwards: 68
From: «+79262719630» ;tag=Zv212cXyU6F2B
To: ;tag=as4f0d2223
Call-ID: 337bce93-387a-1234-798b-002590e5052c
CSeq: 86222354 ACK
Content-Length: 0

— (9 headers 0 lines) —
Really destroying SIP dialog ‘337bce93-387a-1234-798b-002590e5052c’ Method: ACK

INVITE sip:s@1.2.3.4:5060;received=1.2.3.4:41394 SIP/2.0
Via: SIP/2.0/UDP 19.2.41.21;rport;branch=z9hG4bK22eeUmjg1KrQg
Route:
Max-Forwards: 68
From: «+79262719630» ;tag=05Ut47D2rF6mQ
To:
Call-ID: 337f7865-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Contact:
User-Agent: MoLie callswitch1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 373
X-FS-Support: update_display,send_info
Remote-Party-ID: «+79262719630» ;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1453088024 1453088025 IN IP4 19.2.41.21
s=FreeSWITCH
c=IN IP4 19.2.41.21
t=0 0
m=audio 28812 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off — — — —
a=ptime:30
m=audio 28812 RTP/AVP 8 0 18 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off — — — —
a=ptime:20

— (18 headers 16 lines) —
Sending to 19.2.41.21:5060 (NAT)
Using INVITE request as basis request — 337f7865-387a-1234-798b-002590e5052c
Found peer ‘Mo_010’ for ‘89262719630’ from 19.2.41.21:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101

Подскажите хотя бы куда копать?

ps ТС — портянки под споллер причте , мотать устал

«SIP/2.0 488 Not acceptable here» error

I am new to MjSip and I use MjUa for creating a client. I want to connect to a asterisk server. it support G.711 but I can not config my app. I use this config:

 media=audio 4000 rtp/avp

but i still get 488 error please help me. how change «MjUa» config file? here is all message log:

INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK2bfdff77 Max-Forwards: 70 To: "Alice" [email protected]:5060> From: "aziz" [email protected]>;tag=350164683297 Call-ID: [email protected] CSeq: 1 INVITE Contact: [email protected]> Expires: 3600 User-Agent: mjsip 1.7 Content-Length: 141 Content-Type: application/sdp v=0 o=157 0 0 IN IP4 192.168.0.57 s=- c=IN IP4 192.168.0.57 t=0 0 m=audio 4000 rtp/avp 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 -----End-of-message----- 1365314026097: 10:23:46.097 Sun 07 Apr 2013, 192.168.0.254:5060/udp (519 bytes) received SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.57:5060;branch=z9hG4bK2bfdff77;received=192.168.0.57;rport=5060 From: "aziz" [email protected]>;tag=350164683297 To: "Alice" [email protected]:5060>;tag=as3f160681 Call-ID: [email protected] CSeq: 1 INVITE Server: FPBX-2.8.1(1.8.11.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6e640e9a" Content-Length: 0 -----End-of-message----- 1365314026107: 10:23:46.107 Sun 07 Apr 2013, 192.168.0.254:5060/udp (326 bytes) sent ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK2bfdff77 Max-Forwards: 70 To: "Alice" [email protected]:5060>;tag=as3f160681 From: "aziz" [email protected]>;tag=350164683297 Call-ID: [email protected] CSeq: 1 ACK User-Agent: mjsip 1.7 Content-Length: 0 -----End-of-message----- 1365314026151: 10:23:46.151 Sun 07 Apr 2013, 192.168.0.254:5060/udp (706 bytes) sent INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK644461b7 Max-Forwards: 70 To: "Alice" [email protected]:5060> From: "aziz" [email protected]>;tag=350164683297 Call-ID: [email protected] CSeq: 2 INVITE Contact: [email protected]> Expires: 3600 User-Agent: mjsip 1.7 Authorization: Digest username="157", realm="asterisk", nonce="6e640e9a", uri /cdn-cgi/l/email-protection" data-cfemail="f4c1c3b4c5cdc6dac5c2ccdac4dac6c1c0">[email protected]:5060", algorithm=MD5, response="84ff5e12b8325a153e09ac2e316f5b1f" Content-Length: 141 Content-Type: application/sdp v=0 o=157 0 0 IN IP4 192.168.0.57 s=- c=IN IP4 192.168.0.57 t=0 0 m=audio 4000 rtp/avp 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 -----End-of-message----- 1365314026152: 10:23:46.152 Sun 07 Apr 2013, 192.168.0.254:5060/udp (450 bytes) received SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.0.57:5060;branch=z9hG4bK644461b7;received=192.168.0.57;rport=5060 From: "aziz" [email protected]>;tag=350164683297 To: "Alice" [email protected]:5060>;tag=as3f160681 Call-ID: [email protected] CSeq: 2 INVITE Server: FPBX-2.8.1(1.8.11.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -----End-of-message----- 1365314026155: 10:23:46.155 Sun 07 Apr 2013, 192.168.0.254:5060/udp (326 bytes) sent ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK644461b7 Max-Forwards: 70 To: "Alice" [email protected]:5060>;tag=as3f160681 From: "aziz" [email protected]>;tag=350164683297 Call-ID: [email protected] CSeq: 2 ACK User-Agent: mjsip 1.7 Content-Length: 0 -----End-of-message----- 

How to fix Error SIP 488 Not Acceptable in Cisco Call Manager

Did you get this SIP error on your Cisco call manager?

You already change all the codec, configured the transcoder, register with the call manager, and everything seems fine. But outgoing calls still give this error.

SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.1.1:5060;received=192.168.1.1;branch=fg5df4bK4F1C77 To: ;tag=5435345-43543 From: ;tag=E435A43-A43 Call-ID: [email protected] CSeq: 101 INVITE Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: Reason: Q.850;cause=65 Content-Length: 0 

Based on the Wikipedia article List of SIP response codes

488 Not Acceptable Here
Some aspect of the session description or the Request-URI is not acceptable, or Codec issue.

If we want to go to more deep, the IETF rfc3261 explain that message:

488 Not Acceptable Here

The response has the same meaning as 606 (Not Acceptable), but only
applies to the specific resource addressed by the Request-URI and the
request may succeed elsewhere.

A message body containing a description of media capabilities MAY be
present in the response, which is formatted according to the Accept
header field in the INVITE (or application/sdp if not present), the
same as a message body in a 200 (OK) response to an OPTIONS request.

Often this is related to codec incompatibilities. For anyone encountering this issue, they should check whether both sides (server and client) have at least one codes they can negotiate.

To fix that, you should check both side configurations; I mean, if you got an error for a SIP trunk, you should check the SIP trunk provider codec and config; if it’s an interconnection in your company, you should check the second server configuration.

For finding and debugging the codec for both sides, you can capture SIP sessions by tools like Wireshark; you can check the codec for each call.

To fix it, you should go to the SIP trunk and look for “MTP Preferred Originating Codec,” and change it. Make sure both sides use the same codec.

Step 1:
Login to your Cisco Unified CM Administration and click on the Device menu.

Step 2:
Click on Trunk

Step 3:
Select your SIP trunk and click on it to change the configuration.

Step 4:
In the SIP trunk configuration, go to the “SIP Information” section and check the value of “MTP Preferred Originating Codec.”

If the problem is still unresolved, there is one more step.
Most SIP providers want Early Offer INVITEs. They use this always to decide on which codec to offer for the calls.

To configure Delayed-Offer to Early-Offer for SIP Audio Calls at the global level, perform the steps in this section.

Step 1:
Open a terminal and connect to your CUCM console.

Step 2:
And enter the following commands:

1. enable 2. configure terminal 3. voice service voip 4. allow-connections sip 5. early-offer forced 6. exit

Do you want to know how your phone system is performing?

If you are looking for a way to monitor the performance of your Cisco Call Manager, then look no further! We provide everything you need in one simple package that will be easy to install and use. There’s no other service like it out there! It’s not just an amazing product but also an incredible experience you can have every day of your life.

  • cisco , Cisco call manager , CUCM , SIP debug , SIP error , sip trunk , UC Clinic

488 Not acceptable here

  • В этой теме 7 ответов, 7 участников, последнее обновление 7 лет, 1 месяц назад сделано bug0r .

Просмотр 8 сообщений — с 1 по 8 (из 8 всего)
25.06.2016 в 20:05 #11850
Сейчас стало 488 при любых звонках. Ещё пару часов назад всё работало. Снова чудит системка?
25.06.2016 в 20:14 #11851

Снова чудит системка ��
Надо снова копить нервные клетки и планировать обновление… Судя по рассылке в астериске исправили много-много ошибок. Сколько привнесли новых не сообщается.

03.07.2016 в 09:24 #11903
Всё понятно ��
Сейчас кстати опять то же самое. 488 и попытки звонков не отображаются в истории.
17.08.2016 в 09:54 #12325

@demon
488 при любых звонках CSipSimple. Сеть занята Zoiper.
Учётные записи активны и подключены, судя по показаниям смартов и самого сервера.

28.08.2016 в 18:13 #12389

У меня сегодня весь день ошибка 488.
До этого сделал сброс телефона Гигасет.
Не пойму, это только у меня такая ошибка или это проблема общая?

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