при звонке на эхотест на 11 -й секунде пропадает звук
по sip логам видно что в обратном адресе указывается не ip адрес модема а ip машины где стоит sip клиент.
эту проблему я устранил отключив sip alg на клиенте настроил stun сервер.
но звук все равно пропадает. лог приведен ниже.
настраиваю в клиенте на sipnet. Таже проблема.
Подскажите как исправить?

<--- SIP read from UDP:109.252.201.137:44686 --->
INVITE sip:100@194.67.28.210 SIP/2.0
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-f7c01dab021225f3-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:74991595789@109.252.201.137:44686>
To: <sip:100@194.67.28.210>
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 236

v=0
o=- 12937665318281250 1 IN IP4 109.252.201.137
s=CounterPath X-Lite 4.0
c=IN IP4 109.252.201.137
t=0 0
m=audio 52850 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
Sending to 109.252.201.137 : 44686 (no NAT)
Using INVITE request as basis request - YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
Found peer '74991595789' for '74991595789' from 109.252.201.137:44686

<--- Reliably Transmitting (NAT) to 109.252.201.137:44686 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-f7c01dab021225f3-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as629928b8
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="32cbc506"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.' in 7360 ms

(Method: INVITE)
asterisk*CLI>
<--- SIP read from UDP:109.252.201.137:44686 --->
ACK sip:100@194.67.28.210 SIP/2.0
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-f7c01dab021225f3-1---d8754z-;rport
Max-Forwards: 70
To: <sip:100@194.67.28.210>;tag=as629928b8
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from UDP:109.252.201.137:44686 --->
INVITE sip:100@194.67.28.210 SIP/2.0
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:74991595789@109.252.201.137:44686>
To: <sip:100@194.67.28.210>
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest

username="74991595789",realm="asterisk",nonce="32cbc506",uri="sip:100@194.67.28.210",response="59d0

9e2ba3a8fdb18b7207e39cbd8a40",algorithm=MD5
Content-Length: 236

v=0
o=- 12937665318281250 1 IN IP4 109.252.201.137
s=CounterPath X-Lite 4.0
c=IN IP4 109.252.201.137
t=0 0
m=audio 52850 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (14 headers 10 lines) ---
Sending to 109.252.201.137 : 44686 (NAT)
Using INVITE request as basis request - YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
Found peer '74991595789' for '74991595789' from 109.252.201.137:44686
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0

(nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined -

0x1 (telephone-event)
Peer audio RTP is at port 109.252.201.137:52850
Looking for 100 in from-moderator (domain 194.67.28.210)
list_route: hop: <sip:74991595789@109.252.201.137:44686>

<--- Transmitting (NAT) to 109.252.201.137:44686 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Length: 0


<------------>
Audio is at 194.67.28.210 port 15584
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 109.252.201.137:44686 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as59def01d
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 979002606 979002606 IN IP4 194.67.28.210
s=Asterisk PBX 1.6.2.5-0ubuntu1.1
c=IN IP4 194.67.28.210
t=0 0
m=audio 15584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Dec 24 14:58:01] NOTICE[6304]: channel.c:2960 __ast_read: Dropping incompatible voice frame on

SIP/74991595789-000032a2 of format alaw since our native format has changed to 0x4 (ulaw)
Retransmitting #1 (NAT) to 109.252.201.137:44686:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as59def01d
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 979002606 979002606 IN IP4 194.67.28.210
s=Asterisk PBX 1.6.2.5-0ubuntu1.1
c=IN IP4 194.67.28.210
t=0 0
m=audio 15584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to 109.252.201.137:44686:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as59def01d
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 979002606 979002606 IN IP4 194.67.28.210
s=Asterisk PBX 1.6.2.5-0ubuntu1.1
c=IN IP4 194.67.28.210
t=0 0
m=audio 15584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #3 (NAT) to 109.252.201.137:44686:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as59def01d
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 979002606 979002606 IN IP4 194.67.28.210
s=Asterisk PBX 1.6.2.5-0ubuntu1.1
c=IN IP4 194.67.28.210
t=0 0
m=audio 15584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #4 (NAT) to 109.252.201.137:44686:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as59def01d
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 979002606 979002606 IN IP4 194.67.28.210
s=Asterisk PBX 1.6.2.5-0ubuntu1.1
c=IN IP4 194.67.28.210
t=0 0
m=audio 15584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #5 (NAT) to 109.252.201.137:44686:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as59def01d
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 979002606 979002606 IN IP4 194.67.28.210
s=Asterisk PBX 1.6.2.5-0ubuntu1.1
c=IN IP4 194.67.28.210
t=0 0
m=audio 15584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
asterisk*CLI>
<--- SIP read from UDP:109.252.201.137:44686 --->



<------------->
Really destroying SIP dialog 'ZDRmNDQwNWYyMzZkMWRlZGEzMGIwMjgzNDAwMzE4MDI.' Method: REGISTER
Retransmitting #6 (NAT) to 109.252.201.137:44686:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.252.201.137:44686;branch=z9hG4bK-d8754z-65e699c114ff388c-1---

d8754z-;received=109.252.201.137;rport=44686
From: "74991595789"<sip:74991595789@194.67.28.210>;tag=416bf5ca
To: <sip:100@194.67.28.210>;tag=as59def01d
Call-ID: YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:100@194.67.28.210>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 979002606 979002606 IN IP4 194.67.28.210
s=Asterisk PBX 1.6.2.5-0ubuntu1.1
c=IN IP4 194.67.28.210
t=0 0
m=audio 15584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Dec 24 14:58:13] WARNING[990]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on

transmission YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU. for seqno 2 (Critical Response) -- See

doc/sip-retransmit.txt.
[Dec 24 14:58:13] WARNING[990]: chan_sip.c:3806 retrans_pkt: Hanging up call

YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU. - no reply to our critical packet (see doc/sip-

retransmit.txt).
Really destroying SIP dialog 'YjNmNjg0NTQ2NTEwOTI3MzUwOWExYWM0YzhmNTQxMWU.' Method: INVITE