asterisk.ru

Есть GSM транк в локальной сети
Звонок проходит на мобильныи нормально.. после поднятия трубки на мобильном меня слышат а я его не слышу... то есть слышно только в одном направление.
Debug rtp:
Вижу бегут пакеты в обе стороны пример ниже, в чём проблема не могу понять ... если есть время помогите

Sent RTP packet to 10.0.1.241:4042 (type 00, seq 014408, ts 160320, len 000160) -Звонящий
Got RTP packet from 10.0.1.241:4042 (type 00, seq 031294, ts 080320, len 000160)-Звонящий
Sent RTP packet to 10.0.1.225:8000 (type 00, seq 065371, ts 080320, len 000160) - GSM Gate
Got RTP packet from 10.0.1.225:8000 (type 00, seq 000930, ts 160480, len 000160) - GSM Gate

ну наверно проблемы с настройкой GSM гейта ?
так так вы не указали что за модель
не показали обмен инвайтов при общении
помогать пока нечем

как говорил товарищь ded - велком ту пай суппорт

Извините...
Модель : Dinstar DWG2000-1GSM SIP/GSM VoIP Gateway - 1
Обмен инвайтов

Код: Выделить всё

<--- SIP read from UDP:10.0.1.241:60004 --->
INVITE sip:079425423@10.0.1.252 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.241:60004;rport;branch=z9hG4bKPj65756351289447e09e6f90ddf3ef66d3
Max-Forwards: 70
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>
Contact: <sip:2101@10.0.1.241:60004;ob>
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17153 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.9.7
Content-Type: application/sdp
Content-Length: 388

v=0
o=- 3639737599 3639737599 IN IP4 10.0.1.241
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4130 RTP/AVP 8 0 96 3 18 101
c=IN IP4 10.0.1.241
b=TIAS:64000
a=rtcp:4131 IN IP4 10.0.1.241
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 SILK/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (15 headers 19 lines) ---
Sending to 10.0.1.241:60004 (NAT)
Sending to 10.0.1.241:60004 (NAT)
Using INVITE request as basis request - 1e9e17a54cb742f0931e32bcd2369f4c
Found peer '2101' for '2101' from 10.0.1.241:60004

<--- Reliably Transmitting (NAT) to 10.0.1.241:60004 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.1.241:60004;branch=z9hG4bKPj65756351289447e09e6f90ddf3ef66d3;received=10.0.1.241;rport=60004
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as0a07a135
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17153 INVITE
Server: FPBX-AsteriskNOW-12.0.62(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74a360d4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1e9e17a54cb742f0931e32bcd2369f4c' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.0.1.241:60004 --->
ACK sip:079425423@10.0.1.252 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.241:60004;rport;branch=z9hG4bKPj65756351289447e09e6f90ddf3ef66d3
Max-Forwards: 70
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as0a07a135
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17153 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.0.1.241:60004 --->
INVITE sip:079425423@10.0.1.252 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.241:60004;rport;branch=z9hG4bKPj7bf663afc1d742a784e0106313dab272
Max-Forwards: 70
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>
Contact: <sip:2101@10.0.1.241:60004;ob>
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17154 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.9.7
Authorization: Digest username="2101", realm="asterisk", nonce="74a360d4", uri="sip:079425423@10.0.1.252", response="2571db48f3f099978d4c9bedab76a128", algorithm=MD5
Content-Type: application/sdp
Content-Length: 388

v=0
o=- 3639737599 3639737599 IN IP4 10.0.1.241
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4130 RTP/AVP 8 0 96 3 18 101
c=IN IP4 10.0.1.241
b=TIAS:64000
a=rtcp:4131 IN IP4 10.0.1.241
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 SILK/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (16 headers 19 lines) ---
Sending to 10.0.1.241:60004 (NAT)
Using INVITE request as basis request - 1e9e17a54cb742f0931e32bcd2369f4c
Found peer '2101' for '2101' from 10.0.1.241:60004
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 96
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found unknown media description format SILK for ID 96
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|ilbc|g729), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.1.241:4130
Looking for 079425423 in TOT (domain 10.0.1.252)
sip_route_dump: route/path hop: <sip:2101@10.0.1.241:60004;ob>

<--- Transmitting (NAT) to 10.0.1.241:60004 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.241:60004;branch=z9hG4bKPj7bf663afc1d742a784e0106313dab272;received=10.0.1.241;rport=60004
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17154 INVITE
Server: FPBX-AsteriskNOW-12.0.62(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:079425423@10.0.1.252:5060>
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 10.0.1.241:60004:
NOTIFY sip:2101@10.0.1.241:60004;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.1.252:5060;branch=z9hG4bK4251025a;rport
Max-Forwards: 70
From: <sip:2101@10.0.1.252>;tag=as53ba7cc0
To: <sip:2101@10.0.1.252>;tag=0f1347fca07c46a793338188fedde2f4
Contact: <sip:2101@10.0.1.252:5060>
Call-ID: d109fa84087046c1b9b0576efb7b9d23
CSeq: 266 NOTIFY
User-Agent: FPBX-AsteriskNOW-12.0.62(13.0.1)
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 524

<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:2101@10.0.1.252">
<pp:person><status>
<ep:activities><ep:busy/></ep:activities>
</status></pp:person>
<note>On the phone</note>
<tuple id="2101">
<contact priority="1">sip:2101@10.0.1.252</contact>
<status><basic>open</basic></status>
</tuple>
</presence>

---

<--- SIP read from UDP:10.0.1.241:60004 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.252:5060;rport=5060;received=10.0.1.252;branch=z9hG4bK4251025a
Call-ID: d109fa84087046c1b9b0576efb7b9d23
From: <sip:2101@10.0.1.252>;tag=as53ba7cc0
To: <sip:2101@10.0.1.252>;tag=0f1347fca07c46a793338188fedde2f4
CSeq: 266 NOTIFY
Contact: <sip:2101@10.0.1.241:60004;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0

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

<--- Transmitting (NAT) to 10.0.1.241:60004 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.241:60004;branch=z9hG4bKPj7bf663afc1d742a784e0106313dab272;received=10.0.1.241;rport=60004
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as501748c0
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17154 INVITE
Server: FPBX-AsteriskNOW-12.0.62(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:079425423@10.0.1.252:5060>
Content-Length: 0


<------------>
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000000, ts 010880, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000001, ts 011040, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000002, ts 011200, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000003, ts 011360, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000004, ts 011520, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000005, ts 011680, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000006, ts 011840, len 000160)

<--- Reliably Transmitting (NAT) to 10.0.1.241:60004 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.241:60004;branch=z9hG4bKPj7bf663afc1d742a784e0106313dab272;received=10.0.1.241;rport=60004
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as501748c0
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17154 INVITE
Server: FPBX-AsteriskNOW-12.0.62(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:079425423@10.0.1.252:5060>
P-Asserted-Identity: "CID:2101" <sip:079425423@10.0.1.252>
Content-Type: application/sdp
Require: timer
Content-Length: 383

v=0
o=root 828026455 828026455 IN IP4 10.0.1.252
s=Asterisk PBX 13.0.1
c=IN IP4 10.0.1.252
t=0 0
m=audio 11322 RTP/AVP 0 8 3 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

<------------>
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000307, ts 060000, len 000160)

<--- SIP read from UDP:10.0.1.241:60004 --->
ACK sip:079425423@10.0.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.241:60004;rport;branch=z9hG4bKPj75bd50cc005843309907b51a1e722c2b
Max-Forwards: 70
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as501748c0
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17154 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.0.1.241:60004 --->
INVITE sip:079425423@10.0.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.241:60004;rport;branch=z9hG4bKPj7f7297e2f858468591dc9b2541c74dde
Max-Forwards: 70
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as501748c0
Contact: <sip:2101@10.0.1.241:60004;ob>
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17155 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 268

v=0
o=- 3639737599 3639737600 IN IP4 10.0.1.241
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4130 RTP/AVP 0 101
c=IN IP4 10.0.1.241
b=TIAS:64000
a=rtcp:4131 IN IP4 10.0.1.241
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 14 lines) ---
Sending to 10.0.1.241:60004 (NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|ilbc|g729), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.1.241:4130

<--- Transmitting (NAT) to 10.0.1.241:60004 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.241:60004;branch=z9hG4bKPj7f7297e2f858468591dc9b2541c74dde;received=10.0.1.241;rport=60004
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as501748c0
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17155 INVITE
Server: FPBX-AsteriskNOW-12.0.62(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:079425423@10.0.1.252:5060>
Content-Length: 0


<------------>
Audio is at 11322
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec ilbc to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 10.0.1.241:60004 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.241:60004;branch=z9hG4bKPj7f7297e2f858468591dc9b2541c74dde;received=10.0.1.241;rport=60004
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as501748c0
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17155 INVITE
Server: FPBX-AsteriskNOW-12.0.62(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:079425423@10.0.1.252:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 383

v=0
o=root 828026455 828026456 IN IP4 10.0.1.252
s=Asterisk PBX 13.0.1
c=IN IP4 10.0.1.252
t=0 0
m=audio 11322 RTP/AVP 0 8 3 97 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

<------------>

<--- SIP read from UDP:10.0.1.241:60004 --->
ACK sip:079425423@10.0.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.241:60004;rport;branch=z9hG4bKPje7137e812f8749d7b05f9dd22e06fea6
Max-Forwards: 70
From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>;tag=as501748c0
Call-ID: 1e9e17a54cb742f0931e32bcd2369f4c
CSeq: 17155 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Got  RTP packet from    10.0.1.241:4130 (type 00, seq 023006, ts 000160, len 000160)
Sent RTP packet to      10.0.1.225:8000 (type 00, seq 060661, ts 000160, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000308, ts 060160, len 000160)
Sent RTP packet to      10.0.1.241:4130 (type 00, seq 057858, ts 060160, len 000160)
Got  RTP packet from    10.0.1.241:4130 (type 00, seq 023007, ts 000320, len 000160)
Sent RTP packet to      10.0.1.225:8000 (type 00, seq 060662, ts 000320, len 000160)
Got  RTP packet from    10.0.1.225:8000 (type 00, seq 000309, ts 060320, len 000160)
Sent RTP packet to      10.0.1.241:4130 (type 00, seq 057859, ts 060320, len 000160)
Got  RTP packet from    10.0.1.241:4130 (type 00, seq 023008, ts 000480, len 000160)

Подскажите хоть что искать на что обратить внимание.

Советую обратить внимание на конфигурацию NAT. Если GSM шлюз сидит в локалке, следует корректно прописать localnet и nat=never для данного пира.

От себя к себе ?

From: <sip:2101@10.0.1.252>;tag=dfa6989479e54e17be6769d18b45e6e6
To: <sip:079425423@10.0.1.252>

Выключайте на между устройствами

external xx.xxx.226.60

localnet
10.0.1.252/24
10.0.1.0/24
192.168.1.112/30
192.168.0.0/24

Проблемный транк
host=10.0.1.225
allow=ulaw&alaw
type=friend
insecure=very

Готово ... всем спасибо ... проблема была в Microtik .... Сменил IP на GSM Gate и всё заработало.

так нужно было просто
localnet=10.0.1.0/24
localnet=192.168.0.0/16

asterisk.ru

Одностороний звук

Одностороний звук

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Re: Одностороний звук

Re: Одностороний звук

Re: Одностороний звук

Re: Одностороний звук

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