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Asterisk и внешние SIP-клиенты

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Asterisk и внешние SIP-клиенты

Сообщение pavelvlk » 11 авг 2017, 13:57

Приветствую.
Внутри сети есть Asterisk 13 + FreePBX. С внутренними SIP-клиентами все в порядке.
Подключаю внешний SIP-клиент (софтфон на андроиде). Регистрация проходит, звонки идут, а голоса нет в обе стороны.
На границе сети стоит Cisco 2901. NAT настроен, в ACL уже прописано правило permit ip any host <внешний IP Asterisk-а>, а голоса все нет...
В настройках клиента nat включен, транспорт TLS.
Что еще то не хватает?
pavelvlk
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение april22 » 11 авг 2017, 14:56

LocalNet
ExtIP
прописаны ?!
дампы снимали ?!
Своими вопросами , вы загоняете меня в ГУГЛЬ.
april22
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение pavelvlk » 11 авг 2017, 15:11

Прописаны.
Дампы снимал.
Вот при звонке с софтфона:
Код: выделить все
<--- SIP read from TLS:<IP внешнего SIP>:12099 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK3abda1e5;rport=5061;received=<External IP Asterisk>
Contact: <sip:501@<IP внешнего SIP>:12099;transport=TLS>
To: <sip:501@<IP внешнего SIP>:12099;transport=TLS;rinstance=76c169005f8913d5>;tag=1d33a34e
From: "Name" <sip:272@<Local IP Asterisk>>;tag=as34273808
Call-ID: 1ddf693767a10caa37ab298702f30c0a@<Local IP Asterisk>:5061
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rd82a609
Allow-Events: presence, kpml, talk
Content-Length: 328

v=0
o=Zoiper 0 1 IN IP4 10.240.120.89
s=Zoiper
c=IN IP4 10.240.120.89
t=0 0
m=audio 58220 RTP/SAVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:k3Dwj+Hd7xeTxAJElddLtJHsqi1/83frGmGZnnSu
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g729), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.240.120.89:58220
sip_route_dump: route/path hop: <sip:501@<IP внешнего SIP>:12099;transport=TLS>
Transmitting (NAT) to <IP внешнего SIP>:12099:
ACK sip:501@<IP внешнего SIP>:12099;transport=TLS SIP/2.0
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK79a88983;rport
Max-Forwards: 70
From: "Name" <sip:272@<Local IP Asterisk>>;tag=as34273808
To: <sip:501@<IP внешнего SIP>:12099;transport=TLS;rinstance=76c169005f8913d5>;tag=1d33a34e
Contact: <sip:272@<Local IP Asterisk>:5061;transport=TLS>
Call-ID: 1ddf693767a10caa37ab298702f30c0a@<Local IP Asterisk>:5061
CSeq: 102 ACK
User-Agent: FPBX-14.0.1rc1.8(13.13.1)
Content-Length: 0


---
Audio is at 15382
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g726 to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to <IP local SIP>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <IP local SIP>:5060;branch=z9hG4bK1149438339;received=<IP local SIP>;rport=5060
From: <sip:272@asterisk.domain.com>;tag=15666803
To: <sip:501@asterisk.domain.com>;tag=as00d78ee6
Call-ID: 1925377011-5060-70@BJC.BGI.BJD.BFC
CSeq: 691 INVITE
Server: FPBX-14.0.1rc1.8(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:501@<Local IP Asterisk>:5060>
P-Asserted-Identity: "Name-2" <sip:501@asterisk.domain.com>
Content-Type: application/sdp
Require: timer
Content-Length: 352

v=0
o=root 992794860 992794860 IN IP4 <Local IP Asterisk>
s=Asterisk PBX 13.13.1
c=IN IP4 <Local IP Asterisk>
t=0 0
m=audio 15382 RTP/AVP 0 8 2 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
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:150
a=sendrecv

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

<--- SIP read from UDP:<IP local SIP>:5060 --->
ACK sip:501@<Local IP Asterisk>:5060 SIP/2.0
Via: SIP/2.0/UDP <IP local SIP>:5060;branch=z9hG4bK341581567;rport
From: <sip:272@asterisk.domain.com>;tag=15666803
To: <sip:501@asterisk.domain.com>;tag=as00d78ee6
Call-ID: 1925377011-5060-70@BJC.BGI.BJD.BFC
CSeq: 691 ACK
Contact: <sip:272@<IP local SIP>:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2135 1.0.9.18
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


Я просто в них ничего не могу понять...
pavelvlk
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение pavelvlk » 11 авг 2017, 16:39

Очередная мертвая тема по данной тематике...
pavelvlk
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение april22 » 11 авг 2017, 16:42

o=Zoiper 0 1 IN IP4 10.240.120.89

Это адрес вашего зоипера , в серой сети провайдера , и о ней ваш <IP внешнего SIP> ни чего не знает
Своими вопросами , вы загоняете меня в ГУГЛЬ.
april22
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение pavelvlk » 11 авг 2017, 16:53

Странно...
Смотрите, Зоипер это софтфон на котором настроен SIP-клиент. И последний у меня регится на астере под другим IP (в дампе как <IP внешнего SIP>).
Это нужно в зоипере ковыряться?
pavelvlk
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение april22 » 11 авг 2017, 17:57

попробуйте в софтофоне прописать stun сервер
Своими вопросами , вы загоняете меня в ГУГЛЬ.
april22
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение pavelvlk » 11 авг 2017, 18:04

О, включил в настройках зоипера STUN и IP в дэбаге у зоипера стал таким же как и у внешнего SIP-а.
Но голос от этого не появился...
pavelvlk
 
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Re: Asterisk и внешние SIP-клиенты

Сообщение pavelvlk » 11 авг 2017, 18:17

Вот новый сип дэбаг при звонке с софтфона на внутренний SIP:
Код: выделить все
<--- SIP read from TLS:<IP внешнего SIP>:33018 --->
INVITE sip:272@<External IP Asterisk>:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---2f6c7fd2caea03fb;rport
Max-Forwards: 70
Contact: <sip:501@<IP внешнего SIP>:29651;transport=TLS>
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rd82a609
Allow-Events: presence, kpml, talk
Content-Length: 819

v=0
o=Zoiper 0 0 IN IP4 <IP внешнего SIP>
s=Zoiper
c=IN IP4 <IP внешнего SIP>
t=0 0
m=audio 2394 RTP/SAVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqZozEh9+ODlPw==
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqZozEh9+ODlPw==
a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqY=
a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqY=
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+
<------------->
--- (13 headers 18 lines) ---
Sending to <IP внешнего SIP>:33018 (NAT)
Sending to <IP внешнего SIP>:33018 (NAT)
Using INVITE request as basis request - ib6BqqXfqngN_gI4qe3x9A..
Found peer '501' for '501' from <IP внешнего SIP>:33018

<--- Reliably Transmitting (NAT) to <IP внешнего SIP>:33018 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---2f6c7fd2caea03fb;received=<IP внешнего SIP>;rport=33018
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>;tag=as42c5b163
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 1 INVITE
Server: FPBX-14.0.1rc1.8(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c9c2e5d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ib6BqqXfqngN_gI4qe3x9A..' in 6400 ms (Method: INVITE)

<--- SIP read from TLS:<IP внешнего SIP>:33018 --->
ACK sip:272@<External IP Asterisk>:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---2f6c7fd2caea03fb;rport
Max-Forwards: 70
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>;tag=as42c5b163
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from TLS:<IP внешнего SIP>:33018 --->
INVITE sip:272@<External IP Asterisk>:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---7422250d89151c25;rport
Max-Forwards: 70
Contact: <sip:501@<IP внешнего SIP>:29651;transport=TLS>
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rd82a609
Authorization: Digest username="501",realm="asterisk",nonce="3c9c2e5d",uri="sip:272@<External IP Asterisk>:5061;transport=TLS",response="41219462242b5c0549e5caee08ddf9d0",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 819

v=0
o=Zoiper 0 0 IN IP4 <IP внешнего SIP>
s=Zoiper
c=IN IP4 <IP внешнего SIP>
t=0 0
m=audio 2394 RTP/SAVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqZozEh9+ODlPw==
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqZozEh9+ODlPw==
a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqY=
a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+AT32kQKMhqY=
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:EyYIMv1h73c0R/ildLoOhYYzPVn9jln7EkPYBt/+
<------------->
--- (14 headers 18 lines) ---
Sending to <IP внешнего SIP>:33018 (NAT)
Using INVITE request as basis request - ib6BqqXfqngN_gI4qe3x9A..
Found peer '501' for '501' from <IP внешнего SIP>:33018
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
[2017-08-11 17:00:24] WARNING[51000][C-00001515]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_256_CM_HMAC_SHA1_80
[2017-08-11 17:00:24] WARNING[51000][C-00001515]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_256_CM_HMAC_SHA1_32
[2017-08-11 17:00:24] WARNING[51000][C-00001515]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_192_CM_HMAC_SHA1_80
[2017-08-11 17:00:24] WARNING[51000][C-00001515]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_192_CM_HMAC_SHA1_32
Capabilities: us - (ulaw|alaw|gsm|g726|g729|g723|g722), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <IP внешнего SIP>:2394
Looking for 272 in from-internal (domain <External IP Asterisk>)
sip_route_dump: route/path hop: <sip:501@<IP внешнего SIP>:29651;transport=TLS>

<--- Transmitting (NAT) to <IP внешнего SIP>:33018 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---7422250d89151c25;received=<IP внешнего SIP>;rport=33018
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 2 INVITE
Server: FPBX-14.0.1rc1.8(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:272@<Local IP Asterisk>:5061;transport=TLS>
Content-Length: 0



<------------>
Scheduling destruction of SIP dialog '0664168e69cb75726276341b5fc54699@10.100.31.10:5060' in 32000 ms (Method: OPTIONS)
Audio is at 10180
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g729 to SDP
Adding codec g723 to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to <IP local SIP>:48719:
INVITE sips:272@<IP local SIP>:5060;transport=tls SIP/2.0
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK02f0d1fd
Max-Forwards: 70
From: "Name" <sip:501@<Local IP Asterisk>>;tag=as2d841e9a
To: <sips:272@<IP local SIP>:5060;transport=tls>
Contact: <sip:501@<Local IP Asterisk>:5061;transport=TLS>
Call-ID: 19700d2d4f3f7ae70a5252aa6f01f59c@<Local IP Asterisk>:5061
CSeq: 102 INVITE
User-Agent: FPBX-14.0.1rc1.8(13.13.1)
Date: Fri, 11 Aug 2017 14:00:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Name" <sip:501@<Local IP Asterisk>>
Content-Type: application/sdp
Content-Length: 534

v=0
o=root 1503983562 1503983562 IN IP4 <Local IP Asterisk>
s=Asterisk PBX 13.13.1
c=IN IP4 <Local IP Asterisk>
t=0 0
m=audio 10180 RTP/SAVP 0 8 3 111 18 4 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WeRaOt8Fi4Fm4UdFldflhNXtIDjRDVIL82TqE0/e

---

<--- Transmitting (NAT) to <IP внешнего SIP>:33018 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---7422250d89151c25;received=<IP внешнего SIP>;rport=33018
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>;tag=as16fbe585
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 2 INVITE
Server: FPBX-14.0.1rc1.8(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:272@<Local IP Asterisk>:5061;transport=TLS>
P-Asserted-Identity: "Name-2" <sip:272@<External IP Asterisk>>
Content-Length: 0


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

<--- SIP read from TLS:<IP local SIP>:48719 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK02f0d1fd
From: "Name" <sip:501@<Local IP Asterisk>>;tag=as2d841e9a
To: <sips:272@<IP local SIP>:5060;transport=tls>
Call-ID: 19700d2d4f3f7ae70a5252aa6f01f59c@<Local IP Asterisk>:5061
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2135 1.0.9.18
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

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

<--- SIP read from TLS:<IP local SIP>:48719 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK02f0d1fd
From: "Name" <sip:501@<Local IP Asterisk>>;tag=as2d841e9a
To: <sips:272@<IP local SIP>:5060;transport=tls>;tag=1776484377
Call-ID: 19700d2d4f3f7ae70a5252aa6f01f59c@<Local IP Asterisk>:5061
CSeq: 102 INVITE
Contact: <sips:272@<IP local SIP>:5060;transport=tls>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2135 1.0.9.18
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sips:272@<IP local SIP>:5060;transport=tls>

<--- Transmitting (NAT) to <IP внешнего SIP>:33018 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---7422250d89151c25;received=<IP внешнего SIP>;rport=33018
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>;tag=as16fbe585
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 2 INVITE
Server: FPBX-14.0.1rc1.8(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:272@<Local IP Asterisk>:5061;transport=TLS>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2ccc71fb5e4d29550179dc834eee402e@<Local IP Asterisk>:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to <IP local SIP>:48719:
OPTIONS sips:272@<IP local SIP>:5060;transport=tls SIP/2.0
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK63a70c88
Max-Forwards: 70
From: "Unknown" <sip:Unknown@<Local IP Asterisk>>;tag=as255d5e08
To: <sips:272@<IP local SIP>:5060;transport=tls>
Contact: <sip:Unknown@<Local IP Asterisk>:5061;transport=TLS>
Call-ID: 30a6318077e5008a7f599fd96f2335cd@<Local IP Asterisk>:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.1rc1.8(13.13.1)
Date: Fri, 11 Aug 2017 14:00:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from TLS:<IP local SIP>:48719 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK63a70c88
From: "Unknown" <sip:Unknown@<Local IP Asterisk>>;tag=as255d5e08
To: <sips:272@<IP local SIP>:5060;transport=tls>;tag=1052169338
Call-ID: 30a6318077e5008a7f599fd96f2335cd@<Local IP Asterisk>:5061
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2135 1.0.9.18
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0




<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '15a6ecff212c32196d9c57e745b6476a@<Local IP Asterisk>:5060' Method: OPTIONS

<--- SIP read from TLS:<IP local SIP>:48719 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK02f0d1fd
From: "Name" <sip:501@<Local IP Asterisk>>;tag=as2d841e9a
To: <sips:272@<IP local SIP>:5060;transport=tls>;tag=1776484377
Call-ID: 19700d2d4f3f7ae70a5252aa6f01f59c@<Local IP Asterisk>:5061
CSeq: 102 INVITE
Contact: <sips:272@<IP local SIP>:5060;transport=tls>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2135 1.0.9.18
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 432

v=0
o=272 8000 8000 IN IP4 <IP local SIP>
s=SIP Call
c=IN IP4 <IP local SIP>
t=0 0
m=audio 5024 RTP/SAVP 0 8 18 9 111 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:FThzOf2RJPKZAYpxTkz61WdetT5rQ2YnvO2XT+sF|2^32
<------------->
--- (12 headers 17 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 111
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 111
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g729|g723|g722), peer - audio=(ulaw|alaw|g722|g729|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726|g729|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <IP local SIP>:5024
sip_route_dump: route/path hop: <sips:272@<IP local SIP>:5060;transport=tls>
Transmitting (no NAT) to <IP local SIP>:5060:
ACK sips:272@<IP local SIP>:5060;transport=tls SIP/2.0
Via: SIP/2.0/TLS <Local IP Asterisk>:5061;branch=z9hG4bK1257c067
Max-Forwards: 70
From: "Name" <sip:501@<Local IP Asterisk>>;tag=as2d841e9a
To: <sips:272@<IP local SIP>:5060;transport=tls>;tag=1776484377
Contact: <sip:501@<Local IP Asterisk>:5061;transport=TLS>
Call-ID: 19700d2d4f3f7ae70a5252aa6f01f59c@<Local IP Asterisk>:5061
CSeq: 102 ACK
User-Agent: FPBX-14.0.1rc1.8(13.13.1)
Content-Length: 0


---
Audio is at 18760
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to <IP внешнего SIP>:33018 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS <IP внешнего SIP>:29651;branch=z9hG4bK-524287-1---7422250d89151c25;received=<IP внешнего SIP>;rport=33018
From: "501"<sip:501@<External IP Asterisk>:5061;transport=TLS>;tag=e4d9eb21
To: <sip:272@<External IP Asterisk>:5061;transport=TLS>;tag=as16fbe585
Call-ID: ib6BqqXfqngN_gI4qe3x9A..
CSeq: 2 INVITE
Server: FPBX-14.0.1rc1.8(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:272@<Local IP Asterisk>:5061;transport=TLS>
P-Asserted-Identity: "Name-2" <sip:272@<External IP Asterisk>>
Content-Type: application/sdp
Content-Length: 386

v=0
o=root 778446681 778446681 IN IP4 <Local IP Asterisk>
s=Asterisk PBX 13.13.1
c=IN IP4 <Local IP Asterisk>
t=0 0
m=audio 18760 RTP/SAVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HHQgFFkcgE1gPrAGBsGlVa/+4kfw77yvsKDfD4YC
pavelvlk
 
Сообщений: 10
Зарегистрирован: 10 авг 2017, 11:23

Re: Asterisk и внешние SIP-клиенты

Сообщение april22 » 11 авг 2017, 18:19

теперь варешарк вам в руки . и иследовать
Своими вопросами , вы загоняете меня в ГУГЛЬ.
april22
 
Сообщений: 2187
Зарегистрирован: 09 июл 2012, 09:47

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