Trixbox и Комстар (МТС)

[ansh]

Всем привет! Помогите разобраться.
Trixbox внутри сети 192.168.0.0
МТС привел свою сеть отдельно 10.60.89.128
Завел и настроил на роутере. Но исходящая связь работает только когда прописываешь в sip.conf externip - ip который выдал МТС, но после этого не слышно голоса у клиентов снаружи.
Дебаг при работающем транке МТС. но не работающих клиентов снаружи.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

SIP Debugging Enabled for IP: 82.204.255.35:5061
Audio is at 10.60.89.130 port 27688
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 82.204.255.35:5061:
INVITE sip:8926XXXXXXX@82.204.255.35:5061 SIP/2.0
Via: SIP/2.0/UDP 10.60.89.130:5060;branch=z9hG4bK79f7b088;rport
Max-Forwards: 70
From: "Andrey" <sip:495XXXXXXX@10.60.89.130>;tag=as0791f834
To: <sip:8926XXXXXXX@82.204.255.35:5061>
Contact: <sip:495XXXXXXX@10.60.89.130>
Call-ID: 6269060017c771ea51764c422be64fb4@10.60.89.130
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Tue, 29 Jan 2013 09:59:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300

v=0
o=root 1936789659 1936789659 IN IP4 10.60.89.130
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 10.60.89.130
t=0 0
m=audio 27688 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
trixbox1*CLI>
<--- SIP read from UDP://82.204.255.35:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.60.89.130:5060;branch=z9hG4bK79f7b088;rport=5060;received=10.60.89.130
From: "Andrey" <sip:495XXXXXXX@10.60.89.130>;tag=as0791f834
To: <sip:8926XXXXXXX@82.204.255.35:5061>
Call-ID: 6269060017c771ea51764c422be64fb4@10.60.89.130
CSeq: 102 INVITE
Server: MERA MVTS3G v.4.3.0-40
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from UDP://82.204.255.35:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.60.89.130:5060;branch=z9hG4bK79f7b088;rport=5060;received=10.60.89.130
From: "Andrey" <sip:495XXXXXXX@10.60.89.130>;tag=as0791f834
To: <sip:8926XXXXXXX@82.204.255.35:5061>;tag=4132013983-3792829033-285262761-1452131365
Call-ID: 6269060017c771ea51764c422be64fb4@10.60.89.130
CSeq: 102 INVITE
Contact: <sip:8926XXXXXXX@82.204.255.35:5061>
Content-Type: application/sdp
Server: MERA MVTS3G v.4.3.0-40
Content-Length: 218

v=0
o=- 1359453590 1359453590 IN IP4 82.204.254.190
s=-
c=IN IP4 82.204.254.190
t=0 0
m=audio 10618 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 82.204.254.190:10618
Scheduling destruction of SIP dialog '6269060017c771ea51764c422be64fb4@10.60.89.130' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 82.204.255.35:5061:
CANCEL sip:8926XXXXXXX@82.204.255.35:5061 SIP/2.0
Via: SIP/2.0/UDP 10.60.89.130:5060;branch=z9hG4bK79f7b088;rport
Max-Forwards: 70
From: "Andrey" <sip:495XXXXXXX@10.60.89.130>;tag=as0791f834
To: <sip:8926XXXXXXX@82.204.255.35:5061>
Call-ID: 6269060017c771ea51764c422be64fb4@10.60.89.130
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Content-Length: 0


---
Scheduling destruction of SIP dialog '6269060017c771ea51764c422be64fb4@10.60.89.130' in 32000 ms (Method: INVITE)
trixbox1*CLI>
<--- SIP read from UDP://82.204.255.35:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.89.130:5060;branch=z9hG4bK79f7b088;rport=5060;received=10.60.89.130
From: "Andrey" <sip:495XXXXXXX@10.60.89.130>;tag=as0791f834
To: <sip:8926XXXXXXX@82.204.255.35:5061>;tag=4132013983-3792829033-285262761-1452131365
Call-ID: 6269060017c771ea51764c422be64fb4@10.60.89.130
CSeq: 102 CANCEL
Contact: <sip:8926XXXXXXX@82.204.255.35:5061>
Server: MERA MVTS3G v.4.3.0-40
Content-Length: 0


<------------->
-- (9 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from UDP://82.204.255.35:5061 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.60.89.130:5060;branch=z9hG4bK79f7b088;rport=5060;received=10.60.89.130
From: "Andrey" <sip:495XXXXXXX@10.60.89.130>;tag=as0791f834
To: <sip:8926XXXXXXX@82.204.255.35:5061>;tag=4132013983-3792829033-285262761-1452131365
Call-ID: 6269060017c771ea51764c422be64fb4@10.60.89.130
CSeq: 102 INVITE
Server: MERA MVTS3G v.4.3.0-40
Reason: SIP;cause=487;text="Request Terminated"
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 82.204.255.35:5061:
ACK sip:8926XXXXXXX@82.204.255.35:5061 SIP/2.0
Via: SIP/2.0/UDP 10.60.89.130:5060;branch=z9hG4bK79f7b088;rport
Max-Forwards: 70
From: "Andrey" <sip:495XXXXXXX@10.60.89.130>;tag=as0791f834
To: <sip:8926XXXXXXX@82.204.255.35:5061>;tag=4132013983-3792829033-285262761-1452131365
Contact: <sip:495XXXXXXX@10.60.89.130>
Call-ID: 6269060017c771ea51764c422be64fb4@10.60.89.130
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Content-Length: 0


---
Really destroying SIP dialog '6269060017c771ea51764c422be64fb4@10.60.89.130' Method: INVITE
trixbox1*CLI>

транк MTC:
type=friend
host=82.204.255.35
port=5061
fromuser=495XXXXXXX
fromdomain=10.60.89.130
dtmfmode=rfc2833
canreinvite=no
insecure=invite,port
nat=yes
disallow=all
allow=alaw&ulaw
promiscredir=yes

sip.conf:
nat=yes
externip=10.60.89.130
localnet = 192.168.0.0/24

[awsswa]

domain и localnet может быть несколько

localnet=192.168.0.0/255.255.255.0
localnet=10.10.10.0/255.255.255.0

ну и в таком духе.

В вашем случаи вы NAT не там влепили, они не за NATом

[ansh]

прописал nat=no в транке
в sip.conf:
nat=yes
localnet = 192.168.0.0/24
localnet = 10.60.89.128/30
;externip=10.60.89.130

убрал externip и перестал проходить звонок через МТС

[awsswa]

extenip оставьте как было, и localnet не надо вносить ваш внешний адресс

что у вас в domain ?

[ded]

Если укажите в Астериске
bindaddr=0.0.0.0
и навесите адрес 10.60.89.130 на туннельный интерфейс Астериска, то всё должно ходить без особых машинерий.
10.60.89.130 - кому сейчас принадлежит физически?

[awsswa]

опа, ded подал правильную мысль
route покажите, у вас похоже маршрут не прописан

[ansh]

10.60.89.130 - адрес выданный МТС. подключен и настроен в роутере на внешний интерфейс.
[root@router ~]# route
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
mvts.voip.comst 10.60.89.129 255.255.255.255 UGH 0 0 0 eth0
mvts.voip.comst 10.60.89.129 255.255.255.255 UGH 0 0 0 eth0
mvts.voip.comst 10.60.89.129 255.255.255.255 UGH 0 0 0 eth0
mvts.voip.comst 10.60.89.129 255.255.255.255 UGH 0 0 0 eth0
10.60.89.128 10.60.89.129 255.255.255.252 UG 0 0 0 eth0
10.60.89.128 * 255.255.255.252 U 0 0 0 eth0
212.XXX.XXX.XXX * 255.255.255.248 U 0 0 0 eth1
85.XXX.XXX.XXX * 255.255.255.224 U 0 0 0 eth0
192.168.0.0 * 255.255.255.0 U 0 0 0 eth2
link-local * 255.255.0.0 U 0 0 0 eth2
default 212.XXX.XXX.XXX 0.0.0.0 UG 0 0 0 eth1

[Vlad1983]

externip=свой-белый
localnet = 192.168.0.0/24
localnet = 10.60.89.128/30
localnet = 82.204.255.35/32 ; какие ещё у mvts.voip.comst также

[ansh]

nat=yes
externip = 10.60.89.130
localnet = 192.168.0.0/24
localnet = 10.60.89.128/30
localnet = 82.204.255.35/32
localnet = 82.204.255.37/32
localnet = 82.204.254.173/32
localnet = 82.204.254.178/32
не звонит через МТС
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK2c10a5df;rport
Contact: <sip:495XXXXXXX@192.168.0.11>


nat=yes
externip = 10.60.89.130
localnet = 192.168.0.0/24
localnet = 82.204.255.35/32
localnet = 82.204.255.37/32
localnet = 82.204.254.173/32
localnet = 82.204.254.178/32
не звонит через МТС
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK2c10a5df;rport
Contact: <sip:495XXXXXXX@192.168.0.11>

[ansh]

да и забыл сказать что 10.60.89.130 висит на вертуальном интерфейсе eth0:3