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Rossi

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О Rossi

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  1. Аналогичная проблема сейчас с этим параметром qualify=yes после включения srvlookup:ZadarmaFax/83792 144.76.42.108 N 5060 UNREACHABLE Кстати, в вики не поменялось пока вроде ничего - обновилось, но все равно по этой инструкции не работает исх связь. Вот лог:
  2. Так а как включается этот srvlookup ? - создание 3-х транков это и есть включение?
  3. Rossi

    Качество связи

    Вообще в последнее время упало качество связи. Какие-то посторонние шумы слышны в трубке. Эхо-тест ещё ничего, но через 888 не особо помогает. Звоним через freepbx (elastix). Направления Москва, мобильные, калужская обл. С другим провайдером слышимость отличная.
  4. На страничке http://wiki.zadarma.com/index.php/Freepbx актуальная информация? Как настраивается во freepbx srvlookup, нужно ли его включать? Если настраивать по wiki, то не работает исходящая связь (про входящую не знаю). Почему в вики отличие настроек asterisk от freepbx - нужно peer или friend указывать? У меня заработало во freepbx (elastix) так (если несколько host то регистрируется на 5.9.108.25; а если один адрес sip.zadarma.com тогда на 144.76.42.108): type=friend username=51139 secret=secret fromuser=51139 fromdomain=sip.zadarma.com host=sipde.zadarma.com&siplv.zadarma.com&sip2.zadarma.com insecure=invite nat=yes canreinvite=no dtmfmode=rfc2833 disallow=all allow=alaw
  5. Rossi

    Количество линий

    Так это ограничение по входящим линиям, или исходящим, или сумма?
  6. Rossi

    Входящие

    А номера эти 39760 и 60888 - номера задармы и звонок автоматически попадет на эти экстеншены, если на транке задать один контекст или ещё надо где-то настроить? - upd. Да, так работает, номера эти надо занести в поле contact в файле users.confТогда встает вопрос как отлавливать входящие звонки, например отделить номера из Москвы и из-за границы? Т.к., как я понял, теперь эти номера встали на место Pattern'ов В соседней ветке увидел, если задать разные сервера, то тоже работают входящие на разные номера:
  7. Rossi

    Входящие

    А если аккаунты настроены через users.conf (Asterisk GUI)? Там для каждого аккаунта свой контекст создается. В инструкции http://wiki.zadarma.com/index.php/Asterisk что означает contact = 101 как его потом ловить Возможно корявое решение использовать разные адреса кроме sip.zadarma.com на транках, какие альтернативные имена (адреса)?
  8. В какую такую службу поддержки? Звонок, повторяю проходит, но срывается когда берут трубку - занято сразу. Да, это старые логи, вот полные: <--- Transmitting (NAT) to 192.168.1.45:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060 From: <sip:100@192.168.10.30>;tag=18875 To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc Call-ID: 4634 CSeq: 21 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:74955554422@192.168.10.30:5060> Content-Type: application/sdp Content-Length: 274 v=0 o=root 779405766 779405766 IN IP4 192.168.10.30 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 192.168.10.30 t=0 0 m=audio 12670 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:5.9.108.25:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7e96efdd;received=62.205.186.142;rport=28988 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as3086b515 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 102 INVITE Server: Zadarma Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip.zadarma.com", nonce="14f867d0" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing <sip:74955554422@176.9.145.115:5060> for address/port to send to set_destination: set destination to 176.9.145.115:5060 Transmitting (no NAT) to 176.9.145.115:5060: ACK sip:74955554422@176.9.145.115:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7d665df1 Max-Forwards: 70 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Contact: <sip:25638@192.168.10.30:5060> Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- <--- SIP read from UDP:5.9.108.25:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7e96efdd;received=62.205.186.142;rport=28988 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as3086b515 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 102 INVITE Server: Zadarma Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip.zadarma.com", nonce="14f867d0" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing <sip:74955554422@176.9.145.115:5060> for address/port to send to set_destination: set destination to 176.9.145.115:5060 Transmitting (no NAT) to 176.9.145.115:5060: ACK sip:74955554422@176.9.145.115:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7d665df1 Max-Forwards: 70 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Contact: <sip:25638@192.168.10.30:5060> Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- <--- SIP read from UDP:176.9.145.115:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7d665df1;received=62.205.186.142;rport=28988 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 104 INVITE Server: Zadarma Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 324 v=0 o=root 684333766 684333767 IN IP4 176.9.145.115 s=Zadarma Voip c=IN IP4 176.9.145.115 t=0 0 m=audio 11364 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/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 m=video 0 RTP/AVP 99 <-------------> --- (11 headers 15 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 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 GSM for ID 3 Found audio description format telephone-event for ID 101 [Mar 18 22:47:54] WARNING[24299]: chan_sip.c:8949 process_sdp: ignoring 'video' media offer because port number is zeroCapabilities: us - 0x20080e (gsm|ulaw|alaw|g726|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.9.145.115:11364 Peer doesn't provide video list_route: no route [Mar 18 22:47:54] WARNING[24299]: chan_sip.c:13880 __set_address_from_contact: Invalid contact uri (missing sip: or sips:), attempting to use anyway [Mar 18 22:47:54] WARNING[24299]: chan_sip.c:13893 __set_address_from_contact: Invalid URI: parse_uri failed to acquire hostport Transmitting (no NAT) to 176.9.145.115:5060: ACK sip:74955554422@sip.zadarma.com SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK6829626b Max-Forwards: 70 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Contact: <sip:25638@192.168.10.30:5060> Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 --- Reliably Transmitting (no NAT) to 176.9.145.115:5060: BYE sip:74955554422@sip.zadarma.com SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK46a996c9 Max-Forwards: 70 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 105 BYE User-Agent: Asterisk PBX Authorization: Digest username="25638", realm="sip.zadarma.com", algorithm=MD5, uri="sip:74955554422@sip.zadarma.com", nonce="562be498", response="d744ae4624301e41230c42ebf58ab852" X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- Scheduling destruction of SIP dialog '6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com' in 32000 ms (Method: INVITE) -- SIP/Zadarma-00000005 answered SIP/100-00000004 Audio is at 12670 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.45:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060 From: <sip:100@192.168.10.30>;tag=18875 To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc Call-ID: 4634 CSeq: 21 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:74955554422@192.168.10.30:5060> Content-Type: application/sdp Content-Length: 274 v=0 o=root 779405766 779405767 IN IP4 192.168.10.30 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 192.168.10.30 t=0 0 m=audio 12670 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:176.9.145.115:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK46a996c9;received=62.205.186.142;rport=28988 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 105 BYE Server: Zadarma Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 176.9.145.115:5060: BYE sip:74955554422@sip.zadarma.com SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4de0c041 Max-Forwards: 70 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 106 BYE User-Agent: Asterisk PBX Authorization: Digest username="25638", realm="sip.zadarma.com", algorithm=MD5, uri="sip:74955554422@sip.zadarma.com", nonce="562be498", response="d744ae4624301e41230c42ebf58ab852" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/100-00000004' in macro 'trunkdial-failover-0.3' == Spawn extension (DLPN_DialPlan1, 74955554422, 1) exited non-zero on 'SIP/100-00000004' Scheduling destruction of SIP dialog '4634' in 32000 ms (Method: INVITE) <--- SIP read from UDP:176.9.145.115:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4de0c041;received=62.205.186.142;rport=28988 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 106 BYE Server: Zadarma Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com' Method: INVITE Retransmitting #1 (NAT) to 192.168.1.45:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060 From: <sip:100@192.168.10.30>;tag=18875 To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc Call-ID: 4634 CSeq: 21 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:74955554422@192.168.10.30:5060> Content-Type: application/sdp Content-Length: 274 v=0 o=root 779405766 779405767 IN IP4 192.168.10.30 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 192.168.10.30 t=0 0 m=audio 12670 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.1.43:5060 ---> jaK <-------------> Retransmitting #2 (NAT) to 192.168.1.45:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060 From: <sip:100@192.168.10.30>;tag=18875 To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc Call-ID: 4634 CSeq: 21 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:74955554422@192.168.10.30:5060> Content-Type: application/sdp Content-Length: 274 v=0 o=root 779405766 779405767 IN IP4 192.168.10.30 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 192.168.10.30 t=0 0 m=audio 12670 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.1.45:5060 ---> ACK sip:74955554422@192.168.10.30:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.45:5060;rport;branch=z9hG4bK507 From: <sip:100@192.168.10.30>;tag=18875 To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc Call-ID: 4634 CSeq: 21 ACK Contact: <sip:bulat@192.168.1.45> Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing <sip:bulat@19119> for address/port to send to set_destination: set destination to 0.0.74.175:5060 Reliably Transmitting (NAT) to 192.168.1.45:5060: BYE sip:bulat@19119 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4c5c2f2d;rport Max-Forwards: 70 From: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc To: <sip:100@192.168.10.30>;tag=18875 Call-ID: 4634 CSeq: 102 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="100", realm="asterisk", algorithm=MD5, uri="sip:192.168.10.30", nonce="", response="8d3278878ce074848ab411f87e7c7c60" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '4634' in 32000 ms (Method: ACK) <--- SIP read from UDP:192.168.1.45:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4c5c2f2d;rport=5060 From: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc To: <sip:100@192.168.10.30>;tag=18875 Call-ID: 4634 CSeq: 102 BYE User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '4634' Method: ACK <--- SIP read from UDP:192.168.1.45:5060 ---> jaK <-------------> <--- SIP read from UDP:5.9.108.25:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7e96efdd;received=62.205.186.142;rport=28988 From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad To: <sip:74955554422@sip.zadarma.com>;tag=as3086b515 Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com CSeq: 102 INVITE Server: Zadarma Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip.zadarma.com", nonce="14f867d0" Content-Length: 0 <------------->
  9. Здравствуйте! Проблема с совершением исходящих звонков через Zadarma, звонок проходит и сразу же обрывается - слышно занято в обоих концах. Входящие нормально работают. users.conf почти совпадает с http://wiki.zadarma.com/index.php/Asterisk В логах проскакивала ошибка, но сейчас её уже нет: WARNING[15041] chan_sip.c: Invalid contact uri (missing sip: or sips:), attempting to use anyway [Mar 18 16:25:56] WARNING[15041] chan_sip.c: Invalid URI: parse_uri failed to acquire hostport [Mar 18 16:27:30] NOTICE[15041] chan_sip.c: Call from '1023' (192.168.1.51:5062) to extension '8495' rejected because extension not found in context 'DLPN_zadarDP'. [Mar 18 16:28:10] WARNING[15041] chan_sip.c: Invalid contact uri (missing sip: or sips:), attempting to use anyway [Mar 18 16:28:10] WARNING[15041] chan_sip.c: Invalid URI: parse_uri failed to acquire hostportЧто может быть. В астериске новичок, все настраивалось через GUI-веб интерфейс. С другого провайдера сип работают как входящие так и исходящие.
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