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

[minidisk]

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.

Trunk

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

dtmfmode=rfc2833
type=peer
host=*.*.ru
fromuser=user
fromdomain=*.*.ru
secret=есть
username=user
insecure=port,invite
canreinvite=no
context=from-trunk
qualify=yes
disallow=all
allow=alaw&ulaw&gsm&g729

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

sip set debug

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- 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: <sip:s@1.2.3.4:41394>
Max-Forwards: 68
From: "+79262719630" <sip:89262719630@19.2.41.21>;tag=XaggZpUQ1m3vm
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>
Call-ID: 33783cca-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Contact: <sip:mod_sofia@19.2.41.21:5060>
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" <sip:89262719630@19.2.41.21>;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

<--- Reliably Transmitting (NAT) to 19.2.41.21:5060 --->
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" <sip:89262719630@19.2.41.21>;tag=XaggZpUQ1m3vm
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>;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)

<--- SIP read from UDP:19.2.41.21:5060 --->
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: <sip:s@1.2.3.4:41394>
Max-Forwards: 68
From: "+79262719630" <sip:89262719630@19.2.41.21>;tag=XaggZpUQ1m3vm
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>;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

<--- 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=z9hG4bK1SNNSS1c4a24m
Route: <sip:s@1.2.3.4:41394>
Max-Forwards: 68
From: "+79262719630" <sip:89262719630@19.2.41.21>;tag=Zv212cXyU6F2B
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>
Call-ID: 337bce93-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Contact: <sip:mod_sofia@19.2.41.21:5060>
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" <sip:89262719630@19.2.41.21>;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

<--- Reliably Transmitting (NAT) to 19.2.41.21:5060 --->
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" <sip:89262719630@19.2.41.21>;tag=Zv212cXyU6F2B
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>;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)

<--- SIP read from UDP:19.2.41.21:5060 --->
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: <sip:s@1.2.3.4:41394>
Max-Forwards: 68
From: "+79262719630" <sip:89262719630@19.2.41.21>;tag=Zv212cXyU6F2B
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>;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

<--- 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=z9hG4bK22eeUmjg1KrQg
Route: <sip:s@1.2.3.4:41394>
Max-Forwards: 68
From: "+79262719630" <sip:89262719630@19.2.41.21>;tag=05Ut47D2rF6mQ
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>
Call-ID: 337f7865-387a-1234-798b-002590e5052c
CSeq: 86222354 INVITE
Contact: <sip:mod_sofia@19.2.41.21:5060>
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" <sip:89262719630@19.2.41.21>;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

[ded]

Пир то найден
Found peer 'M_010' for '89262719630' from 19.2.41.21:5060
Но вот экстена s в приёмном контексте (говоря языком Астериска) - нету
From: "+79262719630" <sip:89262719630@19.2.41.21>;tag=XaggZpUQ1m3vm
To: <sip:s@1.2.3.4:5060;received=1.2.3.4:41394>;tag=as1471402a

Ещё версия, что частота квантитизации 60 - много
(alaw:60,ulaw:20,gsm:60,g729:60)
Укажите 20
=alaw:20
=ulaw:20

[Vlad1983]

возможно asterisk не понимает дабл m=audio
или воспринимает только первый где a=ptime:30
у вас такие не разрешены Codec Order : (alaw:60,ulaw:20,gsm:60,g729:60)

[minidisk]

Ещё версия, что частота квантитизации 60 - много
(alaw:60,ulaw:20,gsm:60,g729:60)
Укажите 20
=alaw:20
=ulaw:20

было дефолтно, это я уже игрался (https://wiki.asterisk.org/wiki/display/ ... ketization)

еще заметил... после ребута астера 1ый раз (если никто не звонил) с Мегафона проходит и использует именно G729, а потом болт. Такое впечатление (не знаю как корректно сказать), что кодек тупо или отваливается или система на него почему то в след раз забивает....
debug mode 5 ничего тоже не выявил - пока думаю..........

[minidisk]

Тему можно закрывать - решено.
Правда решение тупое и неинтересное )))))))) Снос астера и установка новой версии - и все работает.

[whoim]

И на какой не работал?