realtime + multifon

[nucleon]

здравствуйте,
настраиваю тестовый-астер в конфигурации Realtime, и вот столкнулся с рядом проблем.
1-я) пытаюсь подключить внешнюю линию от мультифона, вроде как подключил, но звонки внешние не приходят.

2-я) когда пытаюсь звонить из локальной сети, то тут возникают нюансы, или линия недоступна или правила не правильные.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: sip.conf

[general]
context=factory
realm=test
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
language=ru
disallow=all
allow=ulaw
allow=alaw
allow=gsm

type=friend
rtcachefriends = yes
rtcache=yes
alwaysauthreject=yes
register => tcp://7921xxxxxxx@multifon.ru:PASSWORD:7921xxxxxxx@193.201.229.35:5060/multifon
localnet=10.10.10.0/24
localnet=10.10.11.0/24

[multifon]
type=peer
host=193.201.229.35
username=7921xxxxxxx@multifon.ru
secret=PASSWORD
insecure=invite,port
fromdomain=multifon.ru
fromuser=7921xxxxxxx
disallow=all
allow=g729,ulaw
qualify=yes
transport=tcp
;nat=never
nat=yes
dtmfmode=inband
context=incoming
;externip=y.y.y.y
;port=5060
outboundproxy=tcp://193.201.229.35
outboundproxyport=5060
;outboundproxyport=5090

регистрация при этом вроде как есть
sip show peers

Name/username Host Dyn Nat ACL Port Status Realtime
100/Alex 10.10.10.x D 60877 Unmonitored Cached RT
101/Andrey 10.10.10.x D 5065 Unmonitored Cached RT
multifon/7921xxxxxxx@mult 193.201.229.35 N 5060 OK (14 ms)
3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]

в extensions.conf добавленны ссылки на базы правил:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: extensions.conf

[default]
switch => Realtime/@extensions
[factory]
switch => Realtime/@extensions
[menu-choise]
switch => Realtime/@extensions
[incoming]
; мегафон мультифон
switch => Realtime/@extensions

в самой же базе вот такая таблица правил (решил указать все малоли):

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: extensions

1 default 000 1 Answer
2 default 000 2 Playback welcome
3 default 000 3 Goto menu-choise,s,1
4 factory 000 1 Answer
5 factory 000 2 Playback welcome
6 factory 000 3 Goto menu-choise,s,1
7 incoming 000 1 Answer
8 incoming 000 2 Playback welcome
9 incoming 000 3 Goto menu-choise,s,1
10 menu-choise s 1 Background MenuVoiceEntry
11 menu-choise s 2 WaitExten
12 menu-choise 1 1 Playback digits/1
13 menu-choise 1 2 Dial sip/100, 15, m
14 menu-choise 1 4 Playback followme/sorry
15 menu-choise 1 5 Playback redirectmail
16 menu-choise 1 6 Goto 1${MAILBOX_EXISTS(100@factory)}1
17 menu-choise 1 101 Playback not-exist
18 menu-choise 1 102 Goto factory, 000, 1
19 menu-choise 1 111 VoiceMailMain 100@factory
20 menu-choise 1 112 Goto factory,000,1
21 menu-choise 2 1 Playback digits/2
24 menu-choise 2 111 VoiceMailMain 100@factory
25 menu-choise 2 112 Goto factory,000,1
26 menu-choise _1XX/_1XX 1 Playback Dialling
27 menu-choise _1XX/_1XX 2 Dial SIP/${CDR(dst)},10,m
28 menu-choise _1XX/_1XX 3 Playback followme/sorry
29 menu-choise _1XX/_1XX 4 Goto 2${MAILBOX_EXISTS(${CDR(dst)}@factory)}1
30 menu-choise _1XX/_1XX 127 Playback vm-goodbye
31 menu-choise _1XX/_1XX 127 Hangup
32 menu-choise _1XX/_1XX 127 Voicemail ${CDR(dst)}@factory
33 menu-choise _1XX/_1XX 127 Playback vm-goodbye
34 menu-choise _1XX/_1XX 127 Wait 2
35 menu-choise _1XX/_1XX 127 Hangup
36 menu-choise i 1 Playback sory
37 menu-choise i 2 Hangup
38 menu-choise t 1 Playback sory
39 menu-choise t 2 Hangup
50 factory _1XX/_1XX 1 Playback Dialling
51 factory _1XX/_1XX 2 Dial SIP/${CDR(dst)},10,m
52 factory _1XX/_1XX 3 Playback followme/sorry
53 factory _1XX/_1XX 4 Goto 2${MAILBOX_EXISTS(${CDR(dst)}@factory)}1
54 factory _1XX/_1XX 127 Playback vm-goodbye
55 factory _1XX/_1XX 127 Hangup
56 factory _1XX/_1XX 127 Voicemail ${CDR(dst)}@factory
57 factory _1XX/_1XX 127 Playback vm-goodbye
58 factory _1XX/_1XX 127 Wait 2
59 factory _1XX/_1XX 127 Hangup
70 factory 777 1 Wait 2
71 factory 777 2 Record /tmp/welcome:gsm
72 factory 777 3 Wait 1
73 factory 777 4 Playback /tmp/welcome
74 factory 777 5 Wait 2
75 factory 777 6 Hangup
80 factory 001 1 VoiceMailMain
151 factory _XXXXXXXXXXX 1 Dial SIP/multifon/${CDR(dst)},,rT
152 factory _XXXXXXXXXXX 2 Playback followme/sorry
178 factory t 1 Playback sory
179 factory t 2 Hangup
180 factory i 1 Playback sory
181 factory i 2 Hangup
200 incoming 7921xxxxxxx@multifon 1 Answer
201 incoming 7921xxxxxxx@multifon 2 Playback welcome
202 incoming 7921xxxxxxx@multifon 3 Goto menu-choise,s,1

цветом, выделил ключевые моменты

сейчас у меня работают звонки только между локальными пользователями и голосовое меню для них-же
а где ошибся, чтобы работала и внешняя линия непонятно.

[nucleon]

вот попробовал sip set debug ...

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: gebug

sip set debug peer multifon
SIP Debugging Enabled for IP: 193.201.229.35:5060
Really destroying SIP dialog '07c8b251137b749f01a7d88c01f91796@127.0.0.1' Method: REGISTER
== Using SIP RTP CoS mark 5
-- Executing Dial("SIP/100-00000004", "SIP/multifon/8<городской номер>,,rT")
== Using SIP RTP CoS mark 5
-- Called multifon/8<городской номер>
[Aug 18 00:06:02] ERROR[9295]: tcptls.c:354 ast_tcptls_client_start: Unable to connect SIP socket to 0.0.0.0:5060: Connection refused
Reliably Transmitting (NAT) to 193.201.229.35:5060:
OPTIONS sip:193.201.229.35 SIP/2.0
Via: SIP/2.0/TCP <local:ip>:5060;branch=z9hG4bK62c75c32;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<local:ip>>;tag=as3d92e17b
To: <sip:193.201.229.35>
Contact: <sip:asterisk@<local:ip>;transport=TCP>
Call-ID: 7e44eef157a2c30f6a64d0a12fc9adea@<local:ip>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.22
Date: Sat, 17 Aug 2013 20:06:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---

<--- SIP read from TCP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP <local:ip>:5060;received=<адрес провайдера:ip>;branch=z9hG4bK62c75c32;rport=49016
From: "asterisk" <sip:asterisk@multifon.ru>;tag=as3d92e17b
To: <sip:multifon.ru>;tag=aprqngfrt-qaajo910000c6
Call-ID: 7e44eef157a2c30f6a64d0a12fc9adea@<local:ip>
CSeq: 102 OPTIONS
Content-Length: 0

может это подскажет найти проблему

[ded]

tcptls.c:354 ast_tcptls_client_start: Unable to connect SIP socket to 0.0.0.0:5060:

- у Вас не настроен ТСР SIP
и это можно увидеть
CLI> sip show settings

[nucleon]

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: sip show settings

Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm test
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.22
SDP Session Name: Asterisk PBX 1.6.2.22
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Enabled
Qualify Freq : 60000 ms

Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No

Network Settings:
---------------------------
SIP address remapping: Disabled, externip is 0.0.0.0
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Localnet: 10.10.10.0/255.255.255.0
10.10.11.0/255.255.255.0
STUN server: 0.0.0.0:0

Global Signalling Settings:
---------------------------
Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes

Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: factory
Nat: Always
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: ru
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
Forward Detected Loops: Yes

Realtime SIP Settings:
----------------------
Realtime Peers: Yes
Realtime Regs: No
Cache Friends: Yes
Update: Yes
Ignore Reg. Expire: No
Save sys. name: No
Auto Clear: 120 (Disabled)

----

[nucleon]

- у Вас не настроен ТСР SIP

это не оно?
tcpenable=yes

[Vlad1983]

мультифон давно уже по UDP работает

[nucleon]

попробовал убрать настройки tcp:
-- Got SIP response 482 "Loop Detected" back from 127.0.0.1
-- Now forwarding SIP/100-00000004 to 'Local/89*********@incoming' (thanks to SIP/multifon-00000005)
[Aug 18 13:08:39] NOTICE[31310]: chan_local.c:710 local_call: No such extension/context 89*********@incoming while calling Local channel
[Aug 18 13:08:39] NOTICE[31310]: app_dial.c:792 do_forward: Forwarding failed to dial 'Local/89*********@incoming'

аналогично после возврата настроек и включения
tcpenable=yes в разделе [general] файла sip.conf

[zzuz]

Обратитесь в техподдержку Мегафона.
Можете попробывать эту статью http://www.line24.ru/archives/1096 .

[Vlad1983]

в пире [multifon]
username=7921xxxxxxx@multifon.ru
заменить на
username=7921xxxxxxx

в диалплане поправить этот бред
200 incoming 7921xxxxxxx@multifon 1 Answer
201 incoming 7921xxxxxxx@multifon 2 Playback welcome
202 incoming 7921xxxxxxx@multifon 3 Goto menu-choise,s,1
соответственно

[nucleon]

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: debug входящего звонка

<--- Reliably Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 193.201.229.35:5060;branch=z9hG4bK0rqcjk0088o17tsdr6s1.1;received=193.201.229.35
From: sip:7<городской номер>@multifon.ru;tag=CE763246313536413B79DF11
To: sip:<мобильный>@multifon.ru:5060;tag=as15a06d5e
Call-ID: 020426B4E781400005149351@SFESIP3-id1-ext
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.22
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Aug 18 13:27:48] NOTICE[31378]: chan_sip.c:20792 handle_request_invite: Call from '<мобильный>@multifon.ru' to extension 'multifon' rejected because extension not found in context 'incoming'.
Scheduling destruction of SIP dialog '020426B4E781400005149351@SFESIP3-id1-ext' in 6400 ms (Method: INVITE)

<--- SIP read from TCP:193.201.229.35:5060 --->
ACK sip:multifon@<local:ip>;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 193.201.229.35:5060;branch=z9hG4bK0rqcjk0088o17tsdr6s1.1
CSeq: 1 ACK
Max-Forwards: 68
From: sip:7<городской номер>@multifon.ru;tag=CE763246313536413B79DF11
To: <sip:<мобильный>@multifon.ru:5060>;tag=as15a06d5e
Call-ID: 020426B4E781400005149351@SFESIP3-id1-ext
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '020426B4E781400005149351@SFESIP3-id1-ext' Method: INVITE
Really destroying SIP dialog '11ced19005b6180c3a3b31c43d38eaaf@127.0.0.1' Method: REGISTER
Reliably Transmitting (NAT) to 193.201.229.35:5060:
OPTIONS sip:193.201.229.35 SIP/2.0
Via: SIP/2.0/TCP <local:ip>:5060;branch=z9hG4bK0b0988a2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<local:ip>>;tag=as0d69a5e9
To: <sip:193.201.229.35>
Contact: <sip:asterisk@<local:ip>;transport=TCP>
Call-ID: 0a30c86714336b2b5af5044334b26fbb@<local:ip>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.22
Date: Sun, 18 Aug 2013 09:28:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from TCP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP <local:ip>:5060;received=<адрес провайдера:ip>;branch=z9hG4bK0b0988a2;rport=48630
From: "asterisk" <sip:asterisk@multifon.ru>;tag=as0d69a5e9
To: <sip:multifon.ru>;tag=aprqngfrt-khtolp20000c6
Call-ID: 0a30c86714336b2b5af5044334b26fbb@<local:ip>
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
[Aug 18 13:28:33] NOTICE[31377]: chan_sip.c:12063 sip_reregister: -- Re-registration for <мобильный>@multifon.ru@193.201.229.35
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0
Via: SIP/2.0/TCP <local:ip>:5060;branch=z9hG4bK6a2b8ae2;rport
Max-Forwards: 70
From: <sip:<мобильный>@multifon.ru>;tag=as04cd027f
To: <sip:<мобильный>@multifon.ru>
Call-ID: 11ced19005b6180c3a3b31c43d38eaaf@127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX 1.6.2.22
Authorization: Digest username="<мобильный>", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTM3NjgxNzIxMjqEVHo7YPKOoB0m7jkBK7FG", response="1b781a2b72cfd688d4fab22cae9d8ef7", opaque="MTM3NjgxNzIxMjqEVHo7YPKOoB0m7jkBK7FG", qop=auth, cnonce="047dd180", nc=0000000d
Expires: 120
Contact: <sip:multifon@<local:ip>;transport=TCP>
Content-Length: 0


---
Really destroying SIP dialog '0a30c86714336b2b5af5044334b26fbb@<local:ip>' Method: OPTIONS

<--- SIP read from TCP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP <local:ip>:5060;received=<адрес провайдера:ip>;branch=z9hG4bK6a2b8ae2;rport=48630
From: <sip:<мобильный>@multifon.ru>;tag=as04cd027f
To: <sip:<мобильный>@multifon.ru>;tag=BE56324631353641D78ADF11
Call-ID: 11ced19005b6180c3a3b31c43d38eaaf@127.0.0.1
CSeq: 115 REGISTER
Contact: <sip:multifon@<local:ip>;transport=TCP>;expires=90;transport=udp
Contact: <sip:<мобильный>@10.10.10.2:5060;transport=tcp>;expires=155
Supported: path
User-Agent: multifon.ru
Expires: 90
Content-Length: 0
Service-Route: <sip:multifon@193.201.229.35:5060;transport=tcp;lr>

входящий вызов начинает фиксироваться только, если мультифон включен (на 10.10.10.2, на этом же компьютере включен программный телефон)
а ссылки вида
https://sm.megafon.ru/sm/client/routing ... &password=***
не работают 404 ошибка