Дано:
1. Астериск в локалке.
2. НАТ (ограниченный конус)
3. Клиент в интернете (ZyXEL P-2602H)
4. На шлюзе на астериск проброшены порты 5060, 10000-20000.
5. Астериск знает внешний IP, nat=yes.
6. На решение менее суток, поэтому IPSec (единственный, поддерживаемый Зюкселем тип ВПН) не подходит, ибо с ним не знаком.

Результат:
ZyXEL регистрируется, соединение устанавливается, голосовое меню слышу, реального собеседника не слышу, как и он меня.

Вопрос:
Какие параметры мне нужно изменить, чтобы восстановить нормальную слышимость?

Спасибо.

Уточни который сценарий - http://asterisk.ru/knowledgebase/Asterisk+SIP+NAT+solutions
ибо пункт 3. Клиент в интернете (ZyXEL P-2602H) неопределённый.

Asterisk как SIP сервер за NAT, клиент на публичном ИП адресе подключается к Asterisk

Железка-клиент:
http://zyxel.ru/content/catalogue/classifier/43/52/638
Это ДСЛ-модем с сип-адаптером.

Если подключаетесь и слышите IVR, а при соединении с внутренним sip клиентом, который за NAT, то скорей всего canreinvite=no! а у Вас yes! Пошел пить яд... :)

реального собеседника не слышу, как и он меня
CLI>sip show channels во время такого разговора? кодеки?
CLI>sip show peers ?

Более точные ответы базируются на тонком, умном, немного ироничном sip set debug peer ZyXEL P-2602H

betman:Ну если бы было так просто)) с реинвайтом все нормально. Во-первых, явно написано в sip.conf canreinvite=no, во-вторых, Dial(...,tT) уже кое-что значит :)

Кодеки везде g711a. Остальное сейчас организую.

Без настроек тяжело догадаться в чем запара, может быть в пустяке. Ну и дебаг и еще раз дебаг...

sip show peers:
328/328 85.173.XXX.XXX D N 5060 Unmonitored

sip show channels:
sip*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message
192.168.0.41 251 0e57367c415976c 0x8 (alaw) No Tx: ACK
85.173.XXX.XXX 328 356066BED414152 0x8 (alaw) No Rx: ACK



<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INVITE sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKc60ca602c08a4739
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUzN4YDMyYD
Call-ID: 348766BED4141526@85.173.XXX.XXX
CSeq: 1 INVITE
Contact: <sip:328@85.173.XXX.XXX:5060>
Content-Type: application/sdp
Content-Length: 239

v=0
o=328 266577860 266577860 IN IP4 85.173.XXX.XXX
s=-
c=IN IP4 85.173.XXX.XXX
t=0 0
m=audio 50000 RTP/AVP 8 97 0
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:97 G726-16/8000
a=sendrecv
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20

<------------->
--- (10 headers 13 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 85.173.XXX.XXX : 5060 (no NAT)
Using INVITE request as basis request - 348766BED4141526@85.173.XXX.XXX
Found user '328' for '328'
sip*CLI>
<--- Reliably Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKc60ca602c08a4739;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUzN4YDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as6764a310
Call-ID: 348766BED4141526@85.173.XXX.XXX
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="325def85"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '348766BED4141526@85.173.XXX.XXX' in 32000 ms (Method: INVITE)
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
ACK sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKc60ca602c08a4739
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as6764a310
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUzN4YDMyYD
Call-ID: 348766BED4141526@85.173.XXX.XXX
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INVITE sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKce40ae774b025327
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUzN4YDMyYD
Call-ID: 348766BED4141526@85.173.XXX.XXX
CSeq: 2 INVITE
Contact: <sip:328@85.173.XXX.XXX:5060>
Authorization: Digest username="328", realm="asterisk", nonce="325def85", uri="sip:111@92.50.XXX.XXX", response="24e799d616816c4f8d96513a2ccce12e", algorithm=MD5
Content-Type: application/sdp
Content-Length: 239

v=0
o=328 266577860 266577860 IN IP4 85.173.XXX.XXX
s=-
c=IN IP4 85.173.XXX.XXX
t=0 0
m=audio 50000 RTP/AVP 8 97 0
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:97 G726-16/8000
a=sendrecv
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20

<------------->
--- (11 headers 13 lines) ---
Sending to 85.173.XXX.XXX : 5060 (NAT)
Using INVITE request as basis request - 348766BED4141526@85.173.XXX.XXX
Found user '328' for '328'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 0
Peer audio RTP is at port 85.173.XXX.XXX:50000
Found audio description format PCMA for ID 8
Found unknown media description format G726-16 for ID 97
Found audio description format PCMU for ID 0
Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 85.173.XXX.XXX:50000
Looking for 111 in office (domain 92.50.XXX.XXX)
list_route: hop: <sip:328@85.173.XXX.XXX:5060>
sip*CLI>
<--- Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKce40ae774b025327;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUzN4YDMyYD
To: <sip:111@92.50.XXX.XXX>
Call-ID: 348766BED4141526@85.173.XXX.XXX
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:111@192.168.0.99>
Content-Length: 0


<------------>
-- Executing [111@office:1] Goto("SIP/328-b7ac8720", "vmenu,s,1") in new stack
-- Goto (vmenu,s,1)
-- Executing [s@vmenu:1] Answer("SIP/328-b7ac8720", "") in new stack
Audio is at 192.168.0.99 port 12736
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP

<--- Reliably Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKce40ae774b025327;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUzN4YDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as3f04b3a6
Call-ID: 348766BED4141526@85.173.XXX.XXX
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:111@192.168.0.99>
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 1848276768 1848276768 IN IP4 192.168.0.99
s=Asterisk PBX 1.6.0.6
c=IN IP4 192.168.0.99
t=0 0
m=audio 12736 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
ACK sip:111@192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKce40ae774b025327
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as3f04b3a6
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUzN4YDMyYD
Call-ID: 348766BED4141526@85.173.XXX.XXX
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
-- Executing [s@vmenu:2] Ringing("SIP/328-b7ac8720", "") in new stack
-- Executing [s@vmenu:3] Wait("SIP/328-b7ac8720", "2") in new stack
-- Executing [s@vmenu:4] Set("SIP/328-b7ac8720", "TIMEOUT(response)=6") in new stack
-- Response timeout set to 6
-- Executing [s@vmenu:5] BackGround("SIP/328-b7ac8720", "pmsk-hello") in new stack
-- <SIP/328-b7ac8720> Playing 'pmsk-hello.ulaw' (language 'ru')
-- Executing [s@vmenu:6] WaitExten("SIP/328-b7ac8720", "") in new stack

<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INVITE sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKdcbbb5eaa5312317
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 1 INVITE
Contact: <sip:328@85.173.XXX.XXX:5060>
Content-Type: application/sdp
Content-Length: 239

v=0
o=328 266686990 266686990 IN IP4 85.173.XXX.XXX
s=-
c=IN IP4 85.173.XXX.XXX
t=0 0
m=audio 50000 RTP/AVP 8 97 0
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:97 G726-16/8000
a=sendrecv
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20

<------------->
--- (10 headers 13 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 85.173.XXX.XXX : 5060 (no NAT)
Using INVITE request as basis request - 475666BED4141526@85.173.XXX.XXX
Found user '328' for '328'
sip*CLI>
<--- Reliably Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKdcbbb5eaa5312317;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as03ed674a
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41faae7e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '475666BED4141526@85.173.XXX.XXX' in 32000 ms (Method: INVITE)
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
ACK sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKdcbbb5eaa5312317
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as03ed674a
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INVITE sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKf894874e8baa6a39
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 2 INVITE
Contact: <sip:328@85.173.XXX.XXX:5060>
Authorization: Digest username="328", realm="asterisk", nonce="41faae7e", uri="sip:111@92.50.XXX.XXX", response="f764b4df952d65662322e6a46a54bea3", algorithm=MD5
Content-Type: application/sdp
Content-Length: 239

v=0
o=328 266686990 266686990 IN IP4 85.173.XXX.XXX
s=-
c=IN IP4 85.173.XXX.XXX
t=0 0
m=audio 50000 RTP/AVP 8 97 0
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:97 G726-16/8000
a=sendrecv
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20

<------------->
--- (11 headers 13 lines) ---
Sending to 85.173.XXX.XXX : 5060 (NAT)
Using INVITE request as basis request - 475666BED4141526@85.173.XXX.XXX
Found user '328' for '328'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 0
Peer audio RTP is at port 85.173.XXX.XXX:50000
Found audio description format PCMA for ID 8
Found unknown media description format G726-16 for ID 97
Found audio description format PCMU for ID 0
Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 85.173.XXX.XXX:50000
Looking for 111 in office (domain 92.50.XXX.XXX)
list_route: hop: <sip:328@85.173.XXX.XXX:5060>
sip*CLI>
<--- Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKf894874e8baa6a39;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
To: <sip:111@92.50.XXX.XXX>
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:111@192.168.0.99>
Content-Length: 0


<------------>
-- Executing [111@office:1] Goto("SIP/328-b7acf738", "vmenu,s,1") in new stack
-- Goto (vmenu,s,1)
-- Executing [s@vmenu:1] Answer("SIP/328-b7acf738", "") in new stack
Audio is at 192.168.0.99 port 18960
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP

<--- Reliably Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKf894874e8baa6a39;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:111@192.168.0.99>
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 573412239 573412239 IN IP4 192.168.0.99
s=Asterisk PBX 1.6.0.6
c=IN IP4 192.168.0.99
t=0 0
m=audio 18960 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
ACK sip:111@192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKf894874e8baa6a39
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
-- Executing [s@vmenu:2] Ringing("SIP/328-b7acf738", "") in new stack
-- Executing [s@vmenu:3] Wait("SIP/328-b7acf738", "2") in new stack
-- Executing [s@vmenu:4] Set("SIP/328-b7acf738", "TIMEOUT(response)=6") in new stack
-- Response timeout set to 6
-- Executing [s@vmenu:5] BackGround("SIP/328-b7acf738", "pmsk-hello") in new stack
-- <SIP/328-b7acf738> Playing 'pmsk-hello.ulaw' (language 'ru')
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INFO sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bK9641de5305849c6b
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 3 INFO
Content-Type: application/dtmf-relay
Content-Length: 27

Signal= 1
Duration= 1000

<------------->
--- (9 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1
sip*CLI>
<--- Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bK9641de5305849c6b;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 3 INFO
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
== CDR updated on SIP/328-b7acf738
-- Executing [1@vmenu:1] Goto("SIP/328-b7acf738", "call-int-num,s,1") in new stack
-- Goto (call-int-num,s,1)
-- Executing [s@call-int-num:1] Set("SIP/328-b7acf738", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing [s@call-int-num:2] Set("SIP/328-b7acf738", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
-- Executing [s@call-int-num:3] BackGround("SIP/328-b7acf738", "pmsk-int-number") in new stack
-- <SIP/328-b7acf738> Playing 'pmsk-int-number.ulaw' (language 'ru')
-- Executing [s@call-int-num:4] WaitExten("SIP/328-b7acf738", "") in new stack
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INFO sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bK0cb4e6a2f490f5bd
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 4 INFO
Content-Type: application/dtmf-relay
Content-Length: 27

Signal= 3
Duration= 1000

<------------->
--- (9 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3
sip*CLI>
<--- Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bK0cb4e6a2f490f5bd;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 4 INFO
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INFO sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKeb116b22883d1921
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 5 INFO
Content-Type: application/dtmf-relay
Content-Length: 27

Signal= 2
Duration= 1000

<------------->
--- (9 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2
sip*CLI>
<--- Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bKeb116b22883d1921;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 5 INFO
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
sip*CLI>
<--- SIP read from UDP://85.173.XXX.XXX:5060 --->
INFO sip:111@92.50.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bK5aa82686664a2ad8
Max-Forwards: 70
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 6 INFO
Content-Type: application/dtmf-relay
Content-Length: 27

Signal= 1
Duration= 1000

<------------->
--- (9 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1
sip*CLI>
<--- Transmitting (NAT) to 85.173.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.173.XXX.XXX:5060;branch=z9hG4bK5aa82686664a2ad8;received=85.173.XXX.XXX
From: <sip:328@92.50.XXX.XXX;user=phone>;tag=xUjN5kDMyYD
To: <sip:111@92.50.XXX.XXX>;tag=as1b5ee7a1
Call-ID: 475666BED4141526@85.173.XXX.XXX
CSeq: 6 INFO
User-Agent: Asterisk PBX 1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
== CDR updated on SIP/328-b7acf738
-- Executing [321@call-int-num:1] NoCDR("SIP/328-b7acf738", "") in new stack
-- Executing [321@call-int-num:2] Dial("SIP/328-b7acf738", "SIP/321,120,tT") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 321
-- SIP/321-081e1160 is ringing

Проблема решена!
была опечатка в externip=92.50.XXX.XXX

Подтверждается моя теория: Стоит запостить вопрос на форум - решение находится само...