Нет исходящих вывозов через asterisk

Добрый день.

Установил, настроил asterisk - подключается к rosnet'у и имеет входящие и исходящие звонки. К астериску подрубаются SIP телефоны и принимают/совершают вызовы.

Все работало нормально на исходящие и на входящие. Но после того как начал ковыряться с кодеками - перестали работать исходящие вызовы, хотя если софтфоном подрубится к роснету - все работает прекрасно.

Не подскажите - где может быть косяк?

Вот конфиги: sip.conf:

[general] register => user:123456@6653055.phone.rosnet.ru/6653055 localnet = 192.168.1.0/255.255.255.0 externip = 87.244.7.206 externrefresh = 60 nat = no canreinvite = no context = 6653055-is language=ru defaultexpiry=600 allowguest = no bindport = 5060 allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 srvlookup = yes subscribecontext = default ;allow = all disallow = all allow = alaw allow = ulaw allow = g729 allow = gsm [sipnet] secret = 123456 defaultuser = user username = user trunkname = sipnet host = 6653055.phone.rosnet.ru context = 6653055-is insecure = invite fromuser = 6653055 fromdomain = 6653055.phone.rosnet.ru type = peer disallow = all allow = alaw allow = ulaw allow = g729 allow = gsm nat = no canreinvite = no dtmfmode = info callcounter = yes 

users.conf:

[general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = yes hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 [1002] type=friend nat = yes canreinvite = no hassip = yes defaultuser = 1002 context=office-users secret=sandello cid_number = 1002 vmsecret=123 host=dynamic dtmfmode=rfc2833 username=1002 disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm 

extensions.conf:

[6653055-is] exten => user,1,Answer exten => user,2,Playback(privetstvie-work) exten => user,3,Dial(SIP/1002) exten => user,4,Hangup [office-users] exten => _XXXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipnet,30,r) 

Звонки извне проходят замечательно, а вот исходящие не идут... Вот логи, при попытке исходящего звонка:

<--- SIP read from UDP:213.108.18.100:60950 ---> INVITE sip:79671983788@87.244.7.206 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK807476705 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206> Call-ID: 527852667@10.0.2.134 CSeq: 1 INVITE Contact: <sip:1002@10.0.2.134:5062> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T18 18.0.14.5 Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 290 v=0 o=- 20006 20006 IN IP4 10.0.2.134 s=SDP data c=IN IP4 10.0.2.134 t=0 0 m=audio 11792 RTP/AVP 8 0 9 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (14 headers 14 lines) --- gateway*CLI> Sending to 10.0.2.134 : 5062 (no NAT) gateway*CLI> Using INVITE request as basis request - 527852667@10.0.2.134 Found peer '1002' for '1002' from 213.108.18.100:60950 <--- Reliably Transmitting (NAT) to 213.108.18.100:60950 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK807476705;received=213.108.18.100 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206>;tag=as1bda2e66 Call-ID: 527852667@10.0.2.134 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4509296c" Content-Length: 0 <------------> gateway*CLI> Scheduling destruction of SIP dialog '527852667@10.0.2.134' in 32000 ms (Method: INVITE) gateway*CLI>  <--- SIP read from UDP:213.108.18.100:60950 ---> ACK sip:79671983788@87.244.7.206 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK807476705 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206>;tag=as1bda2e66 Call-ID: 527852667@10.0.2.134 CSeq: 1 ACK Content-Length: 0 <-------------> gateway*CLI> --- (7 headers 0 lines) --- gateway*CLI>  <--- SIP read from UDP:213.108.18.100:60950 ---> INVITE sip:79671983788@87.244.7.206 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206> Call-ID: 527852667@10.0.2.134 CSeq: 2 INVITE Contact: <sip:1002@10.0.2.134:5062> Authorization: Digest username="1002", realm="asterisk", nonce="4509296c", uri="sip:79671983788@87.244.7.206", response="7a447fd38d7a9c25bab04cfe42ca0dc7", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T18 18.0.14.5 Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 290 v=0 o=- 20006 20006 IN IP4 10.0.2.134 s=SDP data c=IN IP4 10.0.2.134 t=0 0 m=audio 11792 RTP/AVP 8 0 9 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (15 headers 14 lines) --- Sending to 213.108.18.100 : 60950 (NAT) Using INVITE request as basis request - 527852667@10.0.2.134 Found peer '1002' for '1002' from 213.108.18.100:60950 gateway*CLI> Found RTP audio format 8 gateway*CLI> Found RTP audio format 0 Found RTP audio format 9 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 audio description format G722 for ID 9 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (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 10.0.2.134:11792 Looking for 79671983788 in office-users (domain 87.244.7.206) gateway*CLI> list_route: hop: <sip:1002@10.0.2.134:5062> gateway*CLI>  <--- Transmitting (NAT) to 213.108.18.100:60950 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524;received=213.108.18.100 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206> Call-ID: 527852667@10.0.2.134 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671983788@87.244.7.206> Content-Length: 0 <------------> gateway*CLI> Audio is at 87.244.7.206 port 19186 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP gateway*CLI> Reliably Transmitting (no NAT) to 195.90.137.30:5060: INVITE sip:79671983788@6653055.phone.rosnet.ru SIP/2.0 Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;rport Max-Forwards: 70 From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4 To: <sip:79671983788@6653055.phone.rosnet.ru> Contact: <sip:6653055@87.244.7.206> Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.20 Date: Fri, 21 Oct 2011 11:13:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 228 v=0 o=root 2014370682 2014370682 IN IP4 87.244.7.206 s=Asterisk PBX 1.6.2.20 c=IN IP4 87.244.7.206 t=0 0 m=audio 19186 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=sendrecv --- gateway*CLI>  <--- Transmitting (NAT) to 213.108.18.100:60950 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524;received=213.108.18.100 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206>;tag=as603a5a56 Call-ID: 527852667@10.0.2.134 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671983788@87.244.7.206> Content-Length: 0 gateway*CLI>  <------------> gateway*CLI>  <--- SIP read from UDP:195.90.137.30:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;received=87.244.7.206;rport=5060 From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4 To: <sip:79671983788@6653055.phone.rosnet.ru>;tag=8KUVKeRbYer8uhjEGUvKpmDJbsn8iAEq Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru CSeq: 102 INVITE Contact: <sip:79671983788@195.90.137.30:5060> Max-Forwards: 70 User-Agent: Svetets CallManager R12642 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO Content-Length: 0 <-------------> --- (11 headers 0 lines) --- gateway*CLI>  <--- SIP read from UDP:195.90.137.30:5060 ---> SIP/2.0 603 Decline Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;received=87.244.7.206;rport=5060 From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4 To: <sip:79671983788@6653055.phone.rosnet.ru>;tag=9cGY5h7SMFgutpbfCIKLFWT1RZWiKDbU Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru CSeq: 102 INVITE Contact: <sip:79671983788@195.90.137.30:5060> Max-Forwards: 70 User-Agent: Svetets CallManager R12642 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO Content-Length: 0 <-------------> --- (11 headers 0 lines) --- gateway*CLI> Transmitting (no NAT) to 195.90.137.30:5060: ACK sip:79671983788@6653055.phone.rosnet.ru SIP/2.0 Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;rport Max-Forwards: 70 From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4 To: <sip:79671983788@6653055.phone.rosnet.ru>;tag=9cGY5h7SMFgutpbfCIKLFWT1RZWiKDbU Contact: <sip:6653055@87.244.7.206> Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.20 Content-Length: 0 gateway*CLI>  gateway*CLI>  --- gateway*CLI>  <--- Reliably Transmitting (NAT) to 213.108.18.100:60950 ---> SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524;received=213.108.18.100 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206>;tag=as603a5a56 Call-ID: 527852667@10.0.2.134 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21 Content-Length: 0 <------------> gateway*CLI>  <--- SIP read from UDP:213.108.18.100:60950 ---> ACK sip:79671983788@87.244.7.206 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524 From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098 To: <sip:79671983788@87.244.7.206>;tag=as603a5a56 Call-ID: 527852667@10.0.2.134 CSeq: 2 ACK Content-Length: 0 <-------------> gateway*CLI> --- (7 headers 0 lines) --- Really destroying SIP dialog '527852667@10.0.2.134' Method: ACK 

Заранее спасибо.