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Everyone is busy/congested не идет вызов через Sip провайдер

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

Модераторы: april22, Zavr2008

Ответить
aquario
Сообщения: 4
Зарегистрирован: 24 ноя 2016, 17:13

Everyone is busy/congested не идет вызов через Sip провайдер

Сообщение aquario »

есть транк pctel1 - подключен, через него не идут вызовы - сразу отбой == Everyone is busy/congested at this time (1:0/0/1)
год назад все работало.....)) потом не использовался...
из debug SIP/2.0 488 Not acceptable here - полагаю тут спотыкается
сам debug:

Код: Выделить всё

- Executing [079261234567@office:1] Dial("SIP/105-00000917", "SIP/pctel1/079261234567,120") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 10140
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.198.162.30:5060:
INVITE sip:079261234567@217.198.162.30:5060 SIP/2.0
Via: SIP/2.0/UDP my_ip:5060;branch=z9hG4bK6506416b
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as19ccd04c
To: <sip:079261234567@217.198.162.30:5060>
Contact: <sip:my_login@my_ip:5060>
Call-ID: 00648b734917a44373522f4c5b59dfb9@217.198.162.30
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.8.0
Date: Wed, 23 Nov 2016 12:37:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1893294316 1893294316 IN IP4 my_ip
s=Asterisk PBX 10.8.0
c=IN IP4 my_ip
t=0 0
m=audio 10140 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/pctel1/079261234567

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP my_ip:5060;branch=z9hG4bK6506416b;rport=5060
From: "Admin" <sip:my_login@217.198.162.30>;tag=as19ccd04c
To: <sip:079261234567@217.198.162.30:5060>;tag=94387f07b30dcb6d8519c673e3cb8435.faff
Call-ID: 00648b734917a44373522f4c5b59dfb9@217.198.162.30
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="217.198.162.30", nonce="WDWM5lg1i7pW+tgH8WxW5oboW3RPMbjT"
Server: kamailio (4.3.4 (i386/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 217.198.162.30:5060:
ACK sip:079261234567@217.198.162.30:5060 SIP/2.0
Via: SIP/2.0/UDP my_ip:5060;branch=z9hG4bK6506416b
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as19ccd04c
To: <sip:079261234567@217.198.162.30:5060>;tag=94387f07b30dcb6d8519c673e3cb8435.faff
Contact: <sip:my_login@my_ip:5060>
Call-ID: 00648b734917a44373522f4c5b59dfb9@217.198.162.30
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.8.0
Content-Length: 0


---
Audio is at 10140
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.198.162.30:5060:
INVITE sip:079261234567@217.198.162.30:5060 SIP/2.0
Via: SIP/2.0/UDP my_ip:5060;branch=z9hG4bK44657872
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as19ccd04c
To: <sip:079261234567@217.198.162.30:5060>
Contact: <sip:my_login@my_ip:5060>
Call-ID: 00648b734917a44373522f4c5b59dfb9@217.198.162.30
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.8.0
Proxy-Authorization: Digest username="my_login@217.198.162.30", realm="217.198.162.30", algorithm=MD5, uri="sip:079261234567@217.198.162.30:5060", nonce="WDWM5lg1i7pW+tgH8WxW5oboW3RPMbjT", response="2ae91000a0437d26009cdba8c8ca1fc5"
Date: Wed, 23 Nov 2016 12:37:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1893294316 1893294317 IN IP4 my_ip
s=Asterisk PBX 10.8.0
c=IN IP4 my_ip
t=0 0
m=audio 10140 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP my_ip:5060;branch=z9hG4bK44657872;rport=5060
From: "Admin" <sip:my_login@217.198.162.30>;tag=as19ccd04c
To: <sip:079261234567@217.198.162.30:5060>
Call-ID: 00648b734917a44373522f4c5b59dfb9@217.198.162.30
CSeq: 103 INVITE
Server: kamailio (4.3.4 (i386/linux))
Content-Length: 0

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

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP my_ip:5060;rport=5060;branch=z9hG4bK44657872
From: "Admin" <sip:my_login@217.198.162.30>;tag=as19ccd04c
To: <sip:079261234567@217.198.162.30:5060>;tag=as6c2d97bf
Call-ID: 00648b734917a44373522f4c5b59dfb9@217.198.162.30
CSeq: 103 INVITE
Server: PCTEL-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 217.198.162.30:5060:
ACK sip:079261234567@217.198.162.30:5060 SIP/2.0
Via: SIP/2.0/UDP my_ip:5060;branch=z9hG4bK44657872
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as19ccd04c
To: <sip:079261234567@217.198.162.30:5060>;tag=as6c2d97bf
Contact: <sip:my_login@my_ip:5060>
Call-ID: 00648b734917a44373522f4c5b59dfb9@217.198.162.30
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.8.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '00648b734917a44373522f4c5b59dfb9@217.198.162.30' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [079261234567@office:2] Hangup("SIP/105-00000917", "") in new stack
  == Spawn extension (office, 079261234567, 2) exited non-zero on 'SIP/105-00000917'
    -- Executing [h@office:1] Verbose("SIP/105-00000917", "HangupSTATUS=CHANUNAVAIL -- CLID="Admin" <105>  --  SOURCE=105") in new stack
HangupSTATUS=CHANUNAVAIL -- CLID=Admin <105>  --  SOURCE=105
    -- Executing [h@office:2] Hangup("SIP/105-00000917", "") in new stack
  == Spawn extension (office, h, 2) exited non-zero on 'SIP/105-00000917'
куда копать? кодеки...?
Вернуться к началу
ded
Сообщения: 15629
Зарегистрирован: 26 авг 2010, 19:00

Re: Everyone is busy/congested не идет вызов через Sip прова

Сообщение ded »

Kamailio говорит - Ваш звонок важен для нас! :)
Ах, вы предлагаете только GSM? Почему, сопсно? Я это не принимаю -
Not acceptable here.

Поствьте alaw & ulaw и всё поедет.
Вернуться к началу
aquario
Сообщения: 4
Зарегистрирован: 24 ноя 2016, 17:13

Re: Everyone is busy/congested не идет вызов через Sip прова

Сообщение aquario »

поставил ulow
Спасибо заработало
Вернуться к началу
aquario
Сообщения: 4
Зарегистрирован: 24 ноя 2016, 17:13

Re: Everyone is busy/congested не идет вызов через Sip прова

Сообщение aquario »

не совсем все заработало... :(
Вызов проходит но звук идет только в одну сторону..., то есть вызываемый абонент не слышит что говорят с asterisk, он звонящий слышит...
Вернуться к началу
ded
Сообщения: 15629
Зарегистрирован: 26 авг 2010, 19:00

Re: Everyone is busy/congested не идет вызов через Sip прова

Сообщение ded »

Изображение
Вернуться к началу
virus_net
Сообщения: 2337
Зарегистрирован: 05 июн 2013, 08:12
Откуда: Москва

Re: Everyone is busy/congested не идет вызов через Sip прова

Сообщение virus_net »

дампайте трафик, смотрите что там c SDP в INVITE и ходит ли RTP после этого. Проверьте firewall, что он точно не блочит порты по которым должен пойти RTP трафик.

P.S. вот так "my_ip" никогда не маскируйте, т.к. получается просто каша. Хотите скрыть IP ? Не вопрос. Заменяйте цифры на цифры, а не на буквы. Не ясно у вас там реальник ? Серый ?
Серые адреса вообще бессмысленно заменять, а реальник можно превратить в 111.111.111.111, 111.222.111.222 и т.п., но никак не в my_ip
мой SIP URI sip:virus_net@asterisk.ru
bitname.ru - Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)
ENUMER - звони бесплатно и напрямую.
Вернуться к началу
aquario
Сообщения: 4
Зарегистрирован: 24 ноя 2016, 17:13

Re: Everyone is busy/congested не идет вызов через Sip прова

Сообщение aquario »

в sip.conf попробовал поиграться с параметрами - поставил NAT=yes - звук пошел в обе стороны...
хоть ip и белый и iptables отключать пробовал....звука не было
debug c NAT=yes

Код: Выделить всё

 -- Executing [981079261412662@office:1] Dial("SIP/105-00000059", "SIP/pctel1/079261412662,60") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 12610
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 217.198.162.30:5060:
INVITE sip:079261412662@217.198.162.30:5060 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.169:5060;branch=z9hG4bK1fe76c90;rport
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>
Contact: <sip:my_login@111.111.111.169:5060>
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.8.0
Date: Thu, 01 Dec 2016 09:59:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 360

v=0
o=root 1288369017 1288369017 IN IP4 111.111.111.169
s=Asterisk PBX 10.8.0
c=IN IP4 111.111.111.169
t=0 0
m=audio 12610 RTP/AVP 0 18 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/pctel1/079261412662

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 111.111.111.169:5060;branch=z9hG4bK1fe76c90;rport=5060;received=111.111.111.169
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=94387f07b30dcb6d8519c673e3cb8435.0db1
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="217.198.162.30", nonce="WD/zslg/8oZP2mrpfZtOHp7aE52gxiXV"
Server: kamailio (4.3.4 (i386/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 217.198.162.30:5060:
ACK sip:079261412662@217.198.162.30:5060 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.169:5060;branch=z9hG4bK1fe76c90;rport
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=94387f07b30dcb6d8519c673e3cb8435.0db1
Contact: <sip:my_login@111.111.111.169:5060>
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.8.0
Content-Length: 0


---
Audio is at 12610
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 217.198.162.30:5060:
INVITE sip:079261412662@217.198.162.30:5060 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.169:5060;branch=z9hG4bK46f05d27;rport
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>
Contact: <sip:my_login@111.111.111.169:5060>
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.8.0
Proxy-Authorization: Digest username="my_login@217.198.162.30", realm="217.198.162.30", algorithm=MD5, uri="sip:079261412662@217.198.162.30:5060", nonce="WD/zslg/8oZP2mrpfZtOHp7aE52gxiXV", response="514e82c53d95aff6eb78336b98961ae1"
Date: Thu, 01 Dec 2016 09:59:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 360

v=0
o=root 1288369017 1288369018 IN IP4 111.111.111.169
s=Asterisk PBX 10.8.0
c=IN IP4 111.111.111.169
t=0 0
m=audio 12610 RTP/AVP 0 18 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 111.111.111.169:5060;branch=z9hG4bK46f05d27;rport=5060;received=111.111.111.169
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 103 INVITE
Server: kamailio (4.3.4 (i386/linux))
Content-Length: 0

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

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 111.111.111.169:5060;received=111.111.111.169;branch=z9hG4bK46f05d27;rport=5060
Record-Route: <sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
Record-Route: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=as4c7361f3
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 103 INVITE
Server: PCTEL-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:079261412662@192.168.1.237:5060>
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
list_route: hop: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>
list_route: hop: <sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
    -- SIP/pctel1-0000005a is ringing

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 111.111.111.169:5060;received=111.111.111.169;branch=z9hG4bK46f05d27;rport=5060
Record-Route: <sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
Record-Route: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=as4c7361f3
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 103 INVITE
Server: PCTEL-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:079261412662@192.168.1.237:5060>
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
list_route: hop: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>
list_route: hop: <sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
    -- SIP/pctel1-0000005a is ringing

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.169:5060;received=111.111.111.169;branch=z9hG4bK46f05d27;rport=5060
Record-Route: <sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
Record-Route: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=as4c7361f3
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 103 INVITE
Server: PCTEL-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:079261412662@192.168.1.237:5060>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 1156499129 1156499129 IN IP4 192.168.1.237
s=PCTEL-PBX
c=IN IP4 192.168.1.237
t=0 0
m=audio 13150 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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=sendrecv
<------------->
--- (14 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
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 telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729|h261), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.237:13150
[2016-12-01 12:59:19] WARNING[19978]: channel.c:5229 set_format: Unable to find a codec translation path from (g729) to (ulaw)
[2016-12-01 12:59:19] WARNING[19978]: channel.c:5229 set_format: Unable to find a codec translation path from (g729) to (ulaw)
list_route: hop: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>
list_route: hop: <sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
set_destination: Parsing <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e> for address/port to send to
set_destination: set destination to 217.198.162.30:5060
Transmitting (NAT) to 217.198.162.30:5060:
ACK sip:079261412662@192.168.1.237:5060 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.169:5060;branch=z9hG4bK02fd8203;rport
Route: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>,<sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=as4c7361f3
Contact: <sip:my_login@111.111.111.169:5060>
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.8.0
Content-Length: 0


---
    -- SIP/pctel1-0000005a answered SIP/105-00000059
    -- Locally bridging SIP/105-00000059 and SIP/pctel1-0000005a
    -- Executing [h@office:1] Verbose("SIP/105-00000059", "HangupSTATUS=ANSWER -- CLID="Admin" <105>  --  SOURCE=105") in new stack
HangupSTATUS=ANSWER -- CLID=Admin <105>  --  SOURCE=105
    -- Executing [h@office:2] Hangup("SIP/105-00000059", "") in new stack
  == Spawn extension (office, h, 2) exited non-zero on 'SIP/105-00000059'
Scheduling destruction of SIP dialog '3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e> for address/port to send to
set_destination: set destination to 217.198.162.30:5060
Reliably Transmitting (NAT) to 217.198.162.30:5060:
BYE sip:079261412662@192.168.1.237:5060 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.169:5060;branch=z9hG4bK5c75458d;rport
Route: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>,<sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
Max-Forwards: 70
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=as4c7361f3
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 104 BYE
User-Agent: Asterisk PBX 10.8.0
Proxy-Authorization: Digest username="my_login@217.198.162.30", realm="217.198.162.30", algorithm=MD5, uri="sip:079261412662@192.168.1.237:5060", nonce="WD/zslg/8oZP2mrpfZtOHp7aE52gxiXV", response="e0b5fa58b97662ab2b7e82d7d09bc2a5"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (office, 981079261412662, 1) exited non-zero on 'SIP/105-00000059'

<--- SIP read from UDP:217.198.162.30:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.169:5060;received=111.111.111.169;branch=z9hG4bK5c75458d;rport=5060
From: "Admin" <sip:my_login@217.198.162.30>;tag=as01b29c2e
To: <sip:079261412662@217.198.162.30:5060>;tag=as4c7361f3
Call-ID: 3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30
CSeq: 104 BYE
Server: PCTEL-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3e2906e00c67a9a0020b6d545f5f97bd@217.198.162.30' Method: INVITE
Really destroying SIP dialog '1d4348ba25030dd43df382fd116fb055@127.0.1.1' Method: REGISTER
111.111.111.169 - мой белый Ip на астере
192.168.1.0 - такой сетки у меня нет..хотя....
Корректно ли все это, или петляет трафик?

sip.conf кусок

Код: Выделить всё


register => my_login:my_pass@217.198.162.30/20019465

[pctel1]
type=friend
secret=my_pass
defaultuser=my_login@217.198.162.30
fromuser=my_login
fromdomain=217.198.162.30
host=217.198.162.30
port=5060
;insecure=invite
insecure=port,invite
qualify=no
dtmfmode=rfc2833
canreinvite=no
nat=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
allow=h261
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virus_net
Сообщения: 2337
Зарегистрирован: 05 июн 2013, 08:12
Откуда: Москва

Re: Everyone is busy/congested не идет вызов через Sip прова

Сообщение virus_net »

Во-первых, уберите кашу из предлагаемых вами кодеков в сторону прова. Оставьте alaw и ulaw.
Во-вторых:
aquario писал(а):Server: kamailio (4.3.4 (i386/linux))
А затем:
aquario писал(а):Server: PCTEL-PBX
Т.е. у прова на реальнике 217.198.162.30 торчит kamailio, который проксирует трафик до PBX`а, который висит на 192.168.1.30, о чем и говорит:
aquario писал(а):Route: <sip:217.198.162.30;r2=on;lr=on;ftag=as01b29c2e>,<sip:192.168.1.30;r2=on;lr=on;ftag=as01b29c2e>
В-третьих, пров в SDP отдает:
aquario писал(а):o=root 1156499129 1156499129 IN IP4 192.168.1.237
s=PCTEL-PBX
c=IN IP4 192.168.1.237
Что является внутренним IP kamailio. Поэтому задайте вопрос "какого ж..." провайдеру.
мой SIP URI sip:virus_net@asterisk.ru
bitname.ru - Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)
ENUMER - звони бесплатно и напрямую.
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