Trotz, есть очень много способов дебага. Смотрите tcpdump - ходят ли пакеты вообще. Также смотрите дебаг астериска sip set debug peer `ваш_юзер` и rtp debug ip `ip_вашего_юзера`.

Результат команды sip set debug peer 700:

*CLI> sip set debug peer 700
SIP Debugging Enabled for IP: 192.168.0.14:2980
*CLI>

To: "84752539533"<sip:84752539533@192.168.0.17>
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 291

v=0
o=- 4 2 IN IP4 192.168.0.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.14
t=0 0
m=audio 42232 RTP/AVP 107 0 97 8 101
a=alt:1 1 : 9P4yccyL iCdriUzk 192.168.0.14 42232
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 23 18] --- (12 headers 12 lines) ---
[Sep 23 18] == Using SIP RTP TOS bits 184
[Sep 23 18] == Using SIP RTP CoS mark 5
[Sep 23 18] == Using SIP VRTP TOS bits 136
[Sep 23 18] == Using SIP VRTP CoS mark 6
[Sep 23 18] == Using UDPTL TOS bits 184
[Sep 23 18] == Using UDPTL CoS mark 5
[Sep 23 18] Sending to 192.168.0.14 : 2980 (no NAT)
[Sep 23 18] Using INVITE request as basis request - NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
[Sep 23 18] Found peer '700' for '700' from 192.168.0.14:2980
[Sep 23 18]
<--- Reliably Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-d07cc1770370a87d-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as5de8c87d
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 1 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="put.your.domain.here", nonce="1955ae87"
Content-Length: 0


<------------>
[Sep 23 18] Scheduling destruction of SIP dialog 'NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.' in 6528 ms (Method: INVITE)
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
ACK sip:84752539533@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-d07cc1770370a87d-1---d8754z-;rport
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as5de8c87d
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 1 ACK
Content-Length: 0

[size=9]Added after 30 seconds:[/size]

<------------->
[Sep 23 18] --- (7 headers 0 lines) ---
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
INVITE sip:84752539533@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-383ede68e722ca75-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: "84752539533"<sip:84752539533@192.168.0.17>
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="1955ae87",uri="sip:84752539533@192.168.0.17",response="4a2e6449d99e892cb7c691c20ff0562e",algorithm=MD5
Content-Length: 291

v=0
o=- 4 2 IN IP4 192.168.0.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.14
t=0 0
m=audio 42232 RTP/AVP 107 0 97 8 101
a=alt:1 1 : 9P4yccyL iCdriUzk 192.168.0.14 42232
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 23 18] --- (13 headers 12 lines) ---
[Sep 23 18] Sending to 192.168.0.14 : 2980 (NAT)
[Sep 23 18] Using INVITE request as basis request - NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
[Sep 23 18] Found peer '700' for '700' from 192.168.0.14:2980
[Sep 23 18] Found RTP audio format 107
[Sep 23 18] Found RTP audio format 0
[Sep 23 18] Found RTP audio format 97
[Sep 23 18] Found RTP audio format 8
[Sep 23 18] Found RTP audio format 101
[Sep 23 18] Peer audio RTP is at port 192.168.0.14:42232
[Sep 23 18] Found unknown media description format BV32 for ID 107
[Sep 23 18] Found audio description format SPEEX for ID 97
[Sep 23 18] Found audio description format telephone-event for ID 101
[Sep 23 18] Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x20c (ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Sep 23 18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 23 18] Peer audio RTP is at port 192.168.0.14:42232
[Sep 23 18] Looking for 84752539533 in home (domain 192.168.0.17)
[Sep 23 18] list_route: hop: <sip:700@192.168.0.14:2980>
[Sep 23 18]
<--- Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-383ede68e722ca75-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 2 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:84752539533@192.168.0.17>
Content-Length: 0


<------------>
[Sep 23 18] -- Executing [84752539533@home] Set("SIP/700-082d06e0", "CALLERID(all)=”SipPhone” <0021660110>") in new stack

[size=9]Added after 50 seconds:[/size]

[Sep 23 18] Audio is at 192.168.0.17 port 23002
[Sep 23 18] Adding codec 0x4 (ulaw) to SDP
[Sep 23 18] Adding codec 0x8 (alaw) to SDP
[Sep 23 18] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 23 18]
<--- Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-383ede68e722ca75-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 2 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:84752539533@192.168.0.17>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1339671121 1339671121 IN IP4 192.168.0.17
s=Asterisk PBX 1.6.1.6
c=IN IP4 192.168.0.17
t=0 0
m=audio 23002 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=ptime:20
a=sendrecv

<------------>
[Sep 23 18] -- SIP/sipnet-082d49e8 answered SIP/700-082d06e0
[Sep 23 18] Audio is at 192.168.0.17 port 23002
[Sep 23 18] Adding codec 0x4 (ulaw) to SDP
[Sep 23 18] Adding codec 0x8 (alaw) to SDP
[Sep 23 18] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 23 18]
<--- Reliably Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-383ede68e722ca75-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 2 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:84752539533@192.168.0.17>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1339671121 1339671122 IN IP4 192.168.0.17
s=Asterisk PBX 1.6.1.6
c=IN IP4 192.168.0.17
t=0 0
m=audio 23002 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=ptime:20
a=sendrecv

<------------>
Failed to put first frame in the jitterbuffer on channel 'SIP/700-082d06e0'
[Sep 23 18] -- adaptive jitterbuffer created on channel SIP/700-082d06e0
Failed to put first frame in the jitterbuffer on channel 'SIP/sipnet-082d49e8'
[Sep 23 18] -- adaptive jitterbuffer created on channel SIP/sipnet-082d49e8
Resyncing the jb. last_delay 0, this delay -212582544, threshold 1000, new offset 212582544
Resyncing the jb. last_delay 0, this delay -329591, threshold 1000, new offset 329591
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
ACK sip:84752539533@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-9251f816221d834e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 2 ACK
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="1955ae87",uri="sip:84752539533@192.168.0.17",response="4a2e6449d99e892cb7c691c20ff0562e",algorithm=MD5
Content-Length: 0


<------------->
[Sep 23 18] --- (11 headers 0 lines) ---
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
INVITE sip:84752539533@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-154a98205259bf56-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="1955ae87",uri="sip:84752539533@192.168.0.17",response="4a2e6449d99e892cb7c691c20ff0562e",algorithm=MD5
Content-Length: 180

v=0
o=- 4 3 IN IP4 192.168.0.14
s=CounterPath X-Lite 3.0
c=IN IP4 0.0.0.0
t=0 0
m=audio 42232 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly

<------------->
[Sep 23 18] --- (13 headers 9 lines) ---
[Sep 23 18] Sending to 192.168.0.14 : 2980 (NAT)
[Sep 23 18] Found RTP audio format 0
[Sep 23 18] Found RTP audio format 8
[Sep 23 18] Found RTP audio format 101
[Sep 23 18] Peer audio RTP is at port 0.0.0.0:42232
[Sep 23 18] Found audio description format telephone-event for ID 101
[Sep 23 18] Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Sep 23 18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 23 18] Peer audio RTP is at port 0.0.0.0:42232
[Sep 23 18]
<--- Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-154a98205259bf56-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 3 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:84752539533@192.168.0.17>
Content-Length: 0


<------------>
[Sep 23 18] Audio is at 192.168.0.17 port 23002
[Sep 23 18] Adding codec 0x4 (ulaw) to SDP
[Sep 23 18] Adding codec 0x8 (alaw) to SDP
[Sep 23 18] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 23 18]
<--- Reliably Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-154a98205259bf56-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 3 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:84752539533@192.168.0.17>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1339671121 1339671123 IN IP4 192.168.0.17
s=Asterisk PBX 1.6.1.6
c=IN IP4 192.168.0.17
t=0 0
m=audio 23002 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=ptime:20
a=recvonly

<------------>
[Sep 23 18] -- Music class default requested but no musiconhold loaded.
Resyncing the jb. last_delay 0, this delay -212582545, threshold 1000, new offset 212582545
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
ACK sip:84752539533@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-3d77774d3f40c960-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 3 ACK
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="1955ae87",uri="sip:84752539533@192.168.0.17",response="4a2e6449d99e892cb7c691c20ff0562e",algorithm=MD5
Content-Length: 0


<------------->
[Sep 23 18] --- (11 headers 0 lines) ---
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
INVITE sip:84752539533@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-cd744568c47ed717-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 4 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="1955ae87",uri="sip:84752539533@192.168.0.17",response="4a2e6449d99e892cb7c691c20ff0562e",algorithm=MD5
Content-Length: 185

v=0
o=- 4 4 IN IP4 192.168.0.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.14
t=0 0
m=audio 42232 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

[size=9]Added after 30 seconds:[/size]

<------------->
[Sep 23 18] --- (13 headers 9 lines) ---
[Sep 23 18] Sending to 192.168.0.14 : 2980 (NAT)
[Sep 23 18] Found RTP audio format 0
[Sep 23 18] Found RTP audio format 8
[Sep 23 18] Found RTP audio format 101
[Sep 23 18] Peer audio RTP is at port 192.168.0.14:42232
[Sep 23 18] Found audio description format telephone-event for ID 101
[Sep 23 18] Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Sep 23 18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 23 18] Peer audio RTP is at port 192.168.0.14:42232
[Sep 23 18]
<--- Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-cd744568c47ed717-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 4 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:84752539533@192.168.0.17>
Content-Length: 0


<------------>
[Sep 23 18] Audio is at 192.168.0.17 port 23002
[Sep 23 18] Adding codec 0x4 (ulaw) to SDP
[Sep 23 18] Adding codec 0x8 (alaw) to SDP
[Sep 23 18] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 23 18]
<--- Reliably Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-cd744568c47ed717-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=9d58ed7e
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 4 INVITE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:84752539533@192.168.0.17>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1339671121 1339671124 IN IP4 192.168.0.17
s=Asterisk PBX 1.6.1.6
c=IN IP4 192.168.0.17
t=0 0
m=audio 23002 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=ptime:20
a=sendrecv

<------------>
Resyncing the jb. last_delay 0, this delay -212582543, threshold 1000, new offset 212582543
Resyncing the jb. last_delay 0, this delay -329571, threshold 1000, new offset 329571
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
ACK sip:84752539533@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-35036f3ecd4e0813-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: "84752539533"<sip:84752539533@192.168.0.17>;tag=as616a4607
From: <sip:700@192.168.0.17>;tag=9d58ed7e
Call-ID: NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.
CSeq: 4 ACK
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="1955ae87",uri="sip:84752539533@192.168.0.17",response="4a2e6449d99e892cb7c691c20ff0562e",algorithm=MD5
Content-Length: 0


<------------->
[Sep 23 18] --- (11 headers 0 lines) ---
[Sep 23 18] Reliably Transmitting (NAT) to 192.168.0.14:2980:
OPTIONS sip:700@192.168.0.14:2980;rinstance=3520d32852ae3c0a SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK5a1d9d92;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.17>;tag=as689770c3
To: <sip:700@192.168.0.14:2980;rinstance=3520d32852ae3c0a>
Contact: <sip:asterisk@192.168.0.17>
Call-ID: 16f9a1452c1d78d709ea1b2b41849fab@192.168.0.17
CSeq: 102 OPTIONS
User-Agent: AstPbx.ru
Date: Wed, 23 Sep 2009 14:33:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK5a1d9d92;rport=5060
Contact: <sip:192.168.0.14:2980>
To: <sip:700@192.168.0.14:2980;rinstance=3520d32852ae3c0a>;tag=645bcc4c
From: "asterisk"<sip:asterisk@192.168.0.17>;tag=as689770c3
Call-ID: 16f9a1452c1d78d709ea1b2b41849fab@192.168.0.17
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0


<------------->
[Sep 23 18] --- (12 headers 0 lines) ---
[Sep 23 18] Really destroying SIP dialog '16f9a1452c1d78d709ea1b2b41849fab@192.168.0.17' Method: OPTIONS
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->



<------------->
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->



<------------->
[Sep 23 18] Reliably Transmitting (NAT) to 192.168.0.14:2980:
OPTIONS sip:700@192.168.0.14:2980;rinstance=3520d32852ae3c0a SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK36cc8e8e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.17>;tag=as695487eb
To: <sip:700@192.168.0.14:2980;rinstance=3520d32852ae3c0a>
Contact: <sip:asterisk@192.168.0.17>
Call-ID: 762fa7bf3b0457361afc8900256efff9@192.168.0.17
CSeq: 102 OPTIONS
User-Agent: AstPbx.ru
Date: Wed, 23 Sep 2009 14:34:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK36cc8e8e;rport=5060
Contact: <sip:192.168.0.14:2980>
To: <sip:700@192.168.0.14:2980;rinstance=3520d32852ae3c0a>;tag=e9344e57
From: "asterisk"<sip:asterisk@192.168.0.17>;tag=as695487eb
Call-ID: 762fa7bf3b0457361afc8900256efff9@192.168.0.17
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0


<------------->
[Sep 23 18] --- (12 headers 0 lines) ---
[Sep 23 18] Really destroying SIP dialog '762fa7bf3b0457361afc8900256efff9@192.168.0.17' Method: OPTIONS
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->



<------------->
Correct auth, but based on stale nonce received from '<sip:700@192.168.0.17>;tag=as00024c10'
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
SUBSCRIBE sip:asterisk@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-8841165ed6633513-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: <sip:700@192.168.0.17>;tag=as00024c10
From: <sip:700@192.168.0.17>;tag=001dcd3a
Call-ID: ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.
CSeq: 4 SUBSCRIBE
Expires: 300
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="53b9c812",uri="sip:asterisk@192.168.0.17",response="50ca0a65d50c8252492e8f03983658ef",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
[Sep 23 18] --- (13 headers 0 lines) ---
[Sep 23 18] Found peer '700' for '700' from 192.168.0.14:2980
[Sep 23 18]
<--- Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-8841165ed6633513-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=001dcd3a
To: <sip:700@192.168.0.17>;tag=as00024c10
Call-ID: ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.
CSeq: 4 SUBSCRIBE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="put.your.domain.here", nonce="71cca922", stale=true
Content-Length: 0


<------------>
[Sep 23 18] Scheduling destruction of SIP dialog 'ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.' in 6464 ms (Method: SUBSCRIBE)
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
SUBSCRIBE sip:asterisk@192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-bd6bba6a990daf14-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:700@192.168.0.14:2980>
To: <sip:700@192.168.0.17>;tag=as00024c10
From: <sip:700@192.168.0.17>;tag=001dcd3a
Call-ID: ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.
CSeq: 5 SUBSCRIBE
Expires: 300
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="700",realm="put.your.domain.here",nonce="71cca922",uri="sip:asterisk@192.168.0.17",response="7a54dae3e4bd5986f02e3e1437809892",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
[Sep 23 18] --- (13 headers 0 lines) ---
[Sep 23 18] Found peer '700' for '700' from 192.168.0.14:2980
[Sep 23 18] Scheduling destruction of SIP dialog 'ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.' in 310000 ms (Method: SUBSCRIBE)
[Sep 23 18]
<--- Transmitting (NAT) to 192.168.0.14:2980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-bd6bba6a990daf14-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=001dcd3a
To: <sip:700@192.168.0.17>;tag=as00024c10
Call-ID: ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.
CSeq: 5 SUBSCRIBE
Server: AstPbx.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:asterisk@192.168.0.17>;expires=300
Content-Length: 0


<------------>
[Sep 23 18] Reliably Transmitting (NAT) to 192.168.0.14:2980:
NOTIFY sip:700@192.168.0.14:2980 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK49c7c923;rport
Max-Forwards: 70
Route: <sip:700@192.168.0.14:2980>
From: "asterisk" <sip:asterisk@192.168.0.17>;tag=as00024c10
To: <sip:sip:700@192.168.0.14:2980>;tag=001dcd3a
Contact: <sip:asterisk@192.168.0.17>
Call-ID: ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.
CSeq: 103 NOTIFY
User-Agent: AstPbx.ru
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:voicemail@192.168.0.17
Voice-Message: 0/0 (0/0)

---
[Sep 23 18]
<--- SIP read from UDP://192.168.0.14:2980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK49c7c923;rport=5060
Contact: <sip:700@192.168.0.14:2980>
To: <sip:sip:700@192.168.0.14:2980>;tag=001dcd3a
From: "asterisk"<sip:asterisk@192.168.0.17>;tag=as00024c10
Call-ID: ZjQ0NTVkZTlhYTI0NmRiNTBlOTRjYjcwMTZlNzU0YmM.
CSeq: 103 NOTIFY
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0


<------------->
[Sep 23 18] --- (9 headers 0 lines) ---

<------------>
[Sep 23 18] -- adaptive jitterbuffer destroyed on channel SIP/sipnet-082d49e8
[Sep 23 18] == Spawn extension (home, 84752539533, 2) exited non-zero on 'SIP/700-082d06e0'
[Sep 23 18] -- adaptive jitterbuffer destroyed on channel SIP/700-082d06e0
[Sep 23 18] Really destroying SIP dialog 'NzAxYmJkMjdiMWRjZTUwYjczZmQyNmNjMDhlZDI3NzY.' Method: BYE

в начале кусочек потерян, не умещается в буфер

А вот rtp не получается:
*CLI> rtp debug on ip 192.168.0.14
Usage: rtp set debug {on|off|ip host[:port]}
Enable/Disable dumping of all RTP packets. If 'ip' is
specified, limit the dumped packets to those to and from
the specified 'host' with optional port.
The 'rtp debug [off|ip]' command is deprecated and will be removed in a future release. Please use 'rtp set debug {on|off|ip}' instead.
*CLI> rtp set debug ip 192.168.0.14
Lookup failed for 'ip'
Command 'rtp set debug ip 192.168.0.14' failed.
*CLI>

я вот в чтении sdp не копенгаген но я бы смотрел на то в каком порядке кодеки разрешены для xlite.
вообще для начала бы отладил нормальную работу на alaw/ulaw (поставив их первыми в спике разрешенных в xlite) и первыми в allow= в конфигурации пиров в астериске...

Кстати собственно sip.conf стоило бы показать..

Уважаемый!

нет никакого смысла постить дебаг тоннами сюда. Если Вам подсказали как дебажить, то не факт, что всё сюда надо, как на помойку. Старайтесь разбираться самостоятельно. От себя скажу: в предложении кодеков
a=rtpmap:107 BV32/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
нет и намёка на
PCMalaw
PCMulaw

потом появляются
[Sep 23 18] Adding codec 0x4 (ulaw) to SDP
[Sep 23 18] Adding codec 0x8 (alaw) to SDP
видать после нажатия холда?

ещё: у Вас там полно
<--- Transmitting (NAT) to 192.168.0.14:2980 --->
хотя тут же видно по заголовкам, что нет никакого NAT:

Via: SIP/2.0/UDP 192.168.0.14:2980;branch=z9hG4bK-d8754z-bd6bba6a990daf14-1---d8754z-;received=192.168.0.14;rport=2980
From: <sip:700@192.168.0.17>;tag=001dcd3a
To: <sip:700@192.168.0.17>;

За тонну дебага --виноват, такого больше не будет. За терпение к моим ошибкам -спасибо)).
2 dimas -- да в xlite эти кодеки стояли вторыми, и в sip.conf тоже обнаружили косяк в порядке.
2 ded -- спасибо за разъяснение дебага. Статейку-ликбез по дебагу астериска можете мне посоветовать?

Всем спасибо за помощь. Работает)

к сожалению так со всем, с первого раза не вставляет... :)