Проблемы с односторонней слышимостью. Хаотичные.

Односторонняя слышимость.

Избитая проблема. пишу tcpdump-ом звонок. начинаю слушать. и... слышу оба голоса, в оба направления.

Записываю 2 звонка:

оба из города, на один и тот же шлюз, но разные линии(РАР2Т). Один нормально проходит, другой нет.

Т.е. у обоих я в записи слышу голос, но абоненты не слышат голос из города.

пров - * - РАР2Т

дамп снимал м/у * и РАР2Т

Проблема бывает и внутри с локальными звонками. Тогда все пиры находятся в пределах одной сети, одно адресное пространство, никаких натов.

[pap2t](!)
nat=yes
type=friend
host=dynamic
dtmfmode=info
disallow=all
allow=alaw
call-limit=1
deny=0.0.0.0/0.0.0.0
permit=10.1.1.0/255.255.255.0
canreinvite=no
qualify=300
callcounter=yes
callwaiting=no

[133](pap2t)
defaultuser=133
secret=xxx
context=km
callerid="ATC Line 3" <133>

[134](pap2t)
defaultuser=134
secret=yyy
context=km
callerid="ATC Line 4" <134>

С голосом:

|Time     | 10.1.1.111                            |
|         |                   | 10.1.1.236        |                   
|40.966   |         INVITE SDP ( g711A)           |SIP From: sip:111111@10.1.1.111 To:sip:133@10.1.1.236:5060
|         |(5060)   ------------------>  (5060)   |
|40.979   |         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|40.992   |         180 Ringing                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|44.184   |         200 OK SDP ( telephone-event)          |SIP Status
|         |(5060)   <------------------  (5060)   |
|44.184   |         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
|44.191   |         RTP (g711A)                   |RTP Num packets:5338  Duration:106.738s SSRC:0x36223CC1
|         |(17932)  ------------------>  (16398)  |
|44.209   |         RTP (g711A)                   |RTP Num packets:5336  Duration:106.716s SSRC:0x1844FA61
|         |(17932)  <------------------  (16398)  |
|150.945  |         BYE       |                   |SIP Request
|         |(5060)   <------------------  (5060)   |
|150.945  |         200 OK    |                   |SIP Status
|         |(5060)   ------------------>  (5060)   |

Глухой:

|Time     | 10.1.1.111                            |
|         |                   | 10.1.1.236        |                   
|1640.920 |         INVITE SDP ( g711A)           |SIP From: sip:222222@10.1.1.111 To:sip:134@10.1.1.236:5060
|         |(5060)   ------------------>  (5060)   |
|1640.939 |         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|1640.953 |         180 Ringing                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|1644.156 |         200 OK SDP ( telephone-event)          |SIP Status
|         |(5060)   <------------------  (5060)   |
|1644.156 |         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
|1644.161 |         RTP (g711A)                   |RTP Num packets:1670  Duration:33.380s SSRC:0x5A357151
|         |(13556)  ------------------>  (16400)  |
|1644.183 |         RTP (g711A)                   |RTP Num packets:1669  Duration:33.369s SSRC:0xADAD41E9
|         |(13556)  <------------------  (16400)  |
|1677.561 |         BYE       |                   |SIP Request
|         |(5060)   <------------------  (5060)   |
|1677.562 |         200 OK    |                   |SIP Status
|         |(5060)   ------------------>  (5060)   |


    <--- SIP read from UDP:10.1.1.236:5060 --->
SIP/2.0 200 OK
To: <sip:134@10.1.1.236:5060>;tag=731d88e2d9417038i1
From: "asterisk" <sip:asterisk@10.1.1.111>;tag=as68ff9164
Call-ID: 7a40997337376e701762e9313f17f781@10.1.1.111:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK59d979ba
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7a40997337376e701762e9313f17f781@10.1.1.111:5060' Method: OPTIONS
Audio is at 18356
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (NAT) to 10.1.1.236:5060:
INVITE sip:134@10.1.1.236:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK7f9e7b17;rport
Max-Forwards: 70
From: "8111111111" <sip:111111111@10.1.1.111>;tag=as2844ffd6
To: <sip:134@10.1.1.236:5060>
Contact: <sip:111111111@10.1.1.111:5060>
Call-ID: 14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15.1
Date: Thu, 27 Sep 2012 03:14:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 200

v=0
o=root 76669505 76669505 IN IP4 10.1.1.111
s=Asterisk PBX 1.8.15.1
c=IN IP4 10.1.1.111
t=0 0
m=audio 18356 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.1.1.236:5060 --->
SIP/2.0 100 Trying
To: <sip:134@10.1.1.236:5060>
From: "111111111" <sip:111111111@10.1.1.111>;tag=as2844ffd6
Call-ID: 14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK7f9e7b17
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.1.1.236:5060 --->
SIP/2.0 180 Ringing
To: <sip:134@10.1.1.236:5060>;tag=fbe03bfa725de560i1
From: "111111111" <sip:111111111@10.1.1.111>;tag=as2844ffd6
Call-ID: 14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK7f9e7b17
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
list_route: no route
Scheduling destruction of SIP dialog '14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 10.1.1.236:5060:
CANCEL sip:134@10.1.1.236:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK7f9e7b17;rport
Max-Forwards: 70
From: "111111111" <sip:111111111@10.1.1.111>;tag=as2844ffd6
To: <sip:134@10.1.1.236:5060>
Call-ID: 14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.15.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.1.1.236:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:134@10.1.1.236:5060>;tag=fbe03bfa725de560i1
From: "111111111" <sip:111111111@10.1.1.111>;tag=as2844ffd6
Call-ID: 14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK7f9e7b17
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 10.1.1.236:5060:
ACK sip:134@10.1.1.236:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK7f9e7b17;rport
Max-Forwards: 70
From: "111111111" <sip:111111111@10.1.1.111>;tag=as2844ffd6
To: <sip:134@10.1.1.236:5060>;tag=fbe03bfa725de560i1
Contact: <sip:353161@10.1.1.111:5060>
Call-ID: 14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.1.1.236:5060 --->
SIP/2.0 200 OK
To: <sip:134@10.1.1.236:5060>;tag=fbe03bfa725de560i1
From: "111111111" <sip:111111111@10.1.1.111>;tag=as2844ffd6
Call-ID: 14fb0b6d087aa3cc23b7983c51b5766c@10.1.1.111:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 10.1.1.111:5060;branch=z9hG4bK7f9e7b17
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0