Есть оператор городской телефонии, входящие\исходящие работают нормально, НО, если звонить, например, на соторвый, который выключен, то вместо информатора "абонент выключен", получаю гудки, как будто трубку не берёт никто... если же подключиться сип телефоном напрямую к оператору - то всё отлично... поэтому думаю дело в астериске....
вот настройки и лог звонка, подскажите, что подкрутить...
| Цитата: |
| --- -- Called SIP/7654321@provider SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP MY_ASTER_IP:5060;branch=z9hG4bK59aa1601 From: "040" ;tag=as5d93b2db To: ;tag=e2591113c06cc5180f3556e6ad92ea7d.fe86 Call-ID: 511d10c904b95cc9714663022072041a@mydomen CSeq: 102 INVITE Proxy-Authenticate: Digest realm="mydomen", nonce="5024c04a2bb3482a32636237374c660e1d16c1cf" Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to PROV_IP:5060 Transmitting (no NAT) to PROV_IP:5060: ACK sip:7654321@PROV_IP;user=phone SIP/2.0 Via: SIP/2.0/UDP MY_ASTER_IP:5060;branch=z9hG4bK59aa1601 Max-Forwards: 70 From: "040" ;tag=as5d93b2db To: ;tag=e2591113c06cc5180f3556e6ad92ea7d.fe86 Contact: Call-ID: 511d10c904b95cc9714663022072041a@mydomen CSeq: 102 ACK Content-Length: 0 --- Audio is at 65518 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to PROV_IP:5060: INVITE sip:7654321@PROV_IP;user=phone SIP/2.0 Via: SIP/2.0/UDP MY_ASTER_IP:5060;branch=z9hG4bK71966131 Max-Forwards: 70 From: "040" ;tag=as5d93b2db To: Contact: Call-ID: 511d10c904b95cc9714663022072041a@mydomen CSeq: 103 INVITE Proxy-Authorization: Digest username="1234321", realm="mydomen", algorithm=MD5, uri="sip:7654321@PROV_IP;user=phone", nonce="5024c04a2bb3482a32636237374c660e1d16c1cf", response="d011ffa0448abb7bfe63c4b660b51222" Date: Fri, 10 Aug 2012 11:58:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "040" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 284 v=0 o=root 926252293 926252294 IN IP4 MY_ASTER_IP s=DVK phone station c=IN IP4 MY_ASTER_IP t=0 0 m=audio 65518 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP MY_ASTER_IP:5060;branch=z9hG4bK71966131 From: "040" ;tag=as5d93b2db To: Call-ID: 511d10c904b95cc9714663022072041a@mydomen CSeq: 103 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 --- (8 headers 0 lines) --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP MY_ASTER_IP:5060;branch=z9hG4bK71966131 From: "040" ;tag=as5d93b2db To: ;tag=1E330ACC-7A4 Date: Fri, 10 Aug 2012 07:58:22 GMT Call-ID: 511d10c904b95cc9714663022072041a@mydomen Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 261 Session-Expires: 300;refresher=uac Require: timer v=0 o=CiscoSystemsSIP-GW-UserAgent 1055 8457 IN IP4 PROV_VOICE_IP s=SIP Call c=IN IP4 PROV_VOICE_IP t=0 0 m=audio 16710 RTP/AVP 8 101 c=IN IP4 PROV_VOICE_IP a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=direction:active --- (29 headers 0 lines) --- list_route: hop: -- SIP/provider-000001e9 is ringing SIP/2.0 183 Session Progress Via: SIP/2.0/UDP MY_ASTER_IP:5060;branch=z9hG4bK71966131 From: "040" ;tag=as5d93b2db To: ;tag=1E330ACC-7A4 Date: Fri, 10 Aug 2012 07:58:22 GMT Call-ID: 511d10c904b95cc9714663022072041a@mydomen Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 261 Session-Expires: 300;refresher=uac Require: timer v=0 o=CiscoSystemsSIP-GW-UserAgent 1055 8457 IN IP4 PROV_VOICE_IP s=SIP Call c=IN IP4 PROV_VOICE_IP t=0 0 m=audio 16710 RTP/AVP 8 101 c=IN IP4 PROV_VOICE_IP a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=direction:active --- (29 headers 0 lines) --- list_route: hop: -- SIP/provider-000001e9 is ringing |
sip.conf:
| Цитата: |
| register => 1234321:password@PROV_IP/1234321 [provider] trunkname = provider type = friend defaultuser = 1234321 secret = password host = PROV_ID transport=udp insecure = port,invite dtmfmode = auto disallow = all allow = alaw allow = ulaw qualify = yes context = provider-incoming contact = 1234321 fromuser = 1234321 canreinvite = no nat = no language = ru usereqphone = yes videosupport = no session-refresher = uac progressinband=yes prematuremedia=no |
Asterisk 1.8.13.0 на роутере с прошивкой DD-WRT
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Ubuntu 10.04/12.04 - Asterisk 1.8.11.0-rc2/1.8.14.1/1.8.17.0/10.10.0