Всем привет. Почему-то перестали работать исходящие звонки через sipnet. Уже 2-й день бьюсь, но ничего не получается. Сам * висит на "белом" IP. Клиент (X-Lite) находится за NAT. Настройки такие:
[general]
; sipnet.ru
register => 3314073:password@sipnet.ru/3314073
enter code here
[sipnet2]
type=peer
username=user
secret=pass
fromuser=3314073
defaultuser=3314073
callbackextension=3314073
callerid="sipnet" <3314073>
fromdomain=sipnet.ru
host=sipnet.ru
dtmfmode=inband
nat=no
insecure=very
port=5060
context=outgoing-from-sipnet
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=g723
[100]
type=friend
username=100
secret=secret!@#
host=dynamic
nat=yes
qualify=yes
;directmedia=no
call-limit=10
dtmfmode=inband
context=local-phones
disallow=all
allow=alaw
allow=ulaw
[outgoing-from-sipnet]
exten => _9X.,1,Dial(SIP/sipnet2/${EXTEN:1},120)
exten => _9X.,n,Hangup()
[local-phones] include => outgoing-from-sipnet exten => 100,1,Dial(SIP/100) exten => 100,n,Hangup()
pbx*CLI> == Using SIP RTP CoS mark 5 -- Executing [974959815555@local-phones:1] Dial("SIP/101-0000001b", "SIP/sipnet2/74959815555,120") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 212.53.40.40:5060: INVITE sip:74959815555@sipnet.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK40ce6419 Max-Forwards: 70 From: "101" <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060> Contact: <sip:3314073@193.33.185.228:5060> Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.5.0 Date: Tue, 20 Sep 2011 05:30:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 208 v=0 o=root 1364401488 1364401488 IN IP4 193.33.185.228 s=Asterisk PBX 1.8.5.0 c=IN IP4 193.33.185.228 t=0 0 m=audio 16782 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- -- Called SIP/sipnet2/74959815555 <--- SIP read from UDP:212.53.40.40:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK40ce6419 From: "101" <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060> Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 102 INVITE Server: CommuniGatePro/5.4.1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:212.53.40.40:5060 ---> SIP/2.0 401 Authentication required Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK40ce6419 From: "101" <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060>;tag=7FAD969D Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 102 INVITE WWW-Authenticate: Digest realm="etc.tario.ru",nonce="F30A5597CC5A9E9ADD89",opaque="opaqueData",qop="auth",algorithm=MD5 Server: CommuniGatePro/5.4.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 212.53.40.40:5060: ACK sip:74959815555@sipnet.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK40ce6419 Max-Forwards: 70 From: "101" <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060>;tag=7FAD969D Contact: <sip:3314073@193.33.185.228:5060> Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.5.0 Content-Length: 0 --- Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 212.53.40.40:5060: INVITE sip:74959815555@sipnet.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK1ad5c3b7 Max-Forwards: 70 From: "101" <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060> Contact: <sip:3314073@193.33.185.228:5060> Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.5.0 Authorization: Digest username="3314073", realm="etc.tario.ru", algorithm=MD5, uri="sip:74959815555@sipnet.ru:5060", nonce="F30A5597CC5A9E9ADD89", response="969eaa852d244e41298184bb1671ab3b", opaque="opaqueData", qop=auth, cnonce="1660a6b6", nc=00000001 Date: Tue, 20 Sep 2011 05:30:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 208 v=0 o=root 1364401488 1364401489 IN IP4 193.33.185.228 s=Asterisk PBX 1.8.5.0 c=IN IP4 193.33.185.228 t=0 0 m=audio 16782 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:212.53.40.40:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK1ad5c3b7 From: "101" <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060> Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 103 INVITE Server: CommuniGatePro/5.4.1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:212.53.40.40:5060 ---> ****SIP/2.0 403 Forbidden**** Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK1ad5c3b7 From: <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060>;tag=43956a55-84303642 Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 103 INVITE Server: TarioSoftswitch/3.2.12 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 212.53.40.40:5060: ACK sip:74959815555@sipnet.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 193.33.185.228:5060;branch=z9hG4bK1ad5c3b7 Max-Forwards: 70 From: "101" <sip:3314073@sipnet.ru>;tag=as17a59019 To: <sip:74959815555@sipnet.ru:5060>;tag=43956a55-84303642 Contact: <sip:3314073@193.33.185.228:5060> Call-ID: 5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.5.0 Content-Length: 0 --- [Sep 20 09:30:46] WARNING[2425]: chan_sip.c:19568 handle_response_invite: Received response: "Forbidden" from '"101" <sip:3314073@sipnet.ru>;tag=as17a59019' -- SIP/sipnet2-0000001c is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [974959815555@local-phones:2] Hangup("SIP/101-0000001b", "") in new stack == Spawn extension (local-phones, 974959815555, 2) exited non-zero on 'SIP/101-0000001b' Really destroying SIP dialog '5014f7ba3a08a07a5ea81e656e16f943@sipnet.ru' Method: INVITE
В итоге при попытке звонка получаю Forbidden. Почему sipnet посылает?
Блин, а через call файл с самого * звонок проходит. Хрень какая-то!
телефон с именем 100, а звонки с него разве идут? что за цифры 101?
Подскажите мужики по сипнету - с правилами что то запутался. Хочу чтоб допустим набираю +7 или например 44 и номер телефона и звонок шёл через сипнет - тобишь через код. Сипнет вроде зарегился. Просто местные звонки идут через гор. линию и сотовые через goip, а иногда надо например в москву позвонить. Стоит в офисе Elastix 2.3.0
insecure=invite
dtmfmode=rfc2833
host = sipnet.ru
context = from-sipnet
insecure = invite
fromuser = SIPNETID
fromdomain = sipnet.ru
type = peer
disallow = all
allow = alaw
allow = ulaw
canreinvite=no
allow = g729
nat = no
Задан: 2011-09-20 09:35:30 +0400
Просмотрен: 12,923 раз
Обновлен: Jul 20 '12
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.