MyPBX - create_addr: No such host

Question

Здравствуйте. Имеется в наличии MyPBX 1600 v4. На нём установлен астерикс Asterisk SVN--r1826M. Настройка только через веб интерфейс.

Зарегистрировал внешнюю VOIP линию. По статусу вижу что линия зарегистрирована. Залогинился на астериск с другого компьютера софтфоном и начинаю звонить. Да кстати настроил диалплан - все исходящие уходят через созданную VOIP линию.

Ну в общем не работает - сразу идут короткие гудки. Нашел как на это устройство можно залезть через ssh - залез, включил дебаг sip и сделал тестовый звонок. Нашел проблемную строчку, но не знаю как её интерпретировать и как поправить через WEB интерфейс такую ошибку. В общем ошибка вот такая:

[2012-05-06 21:13:42] WARNING[3117]: chan_sip.c:5489 create_addr: No such host: trunk-baza

А вот весь дебаг сипа:

MyPBX*CLI> set sip debug on No such command 'set sip debug on' (type 'core show help set sip' for other possible commands) [2012-05-06 21:13:14] NOTICE[432]: utils.c:305 HostPoolUpdater: ======host:MyPBX update dns====== MyPBX*CLI> sip set debug on SIP Debugging enabled [2012-05-06 21:13:22] NOTICE[432]: utils.c:305 HostPoolUpdater: ======host:sip.qwerty.cnt.ru update dns====== MyPBX*CLI> <--- SIP read from UDP:192.168.1.2:61424 ---> <--- SIP read from UDP:192.168.1.2:61424 ---> INVITE sip:81079030016641@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-66444c0eb6773214-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:501@192.168.1.2:61424> To: "81079030016641"<sip:81079030016641@192.168.1.3> From: "501"<sip:501@192.168.1.3>;tag=3d303e4e Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102u stamp 52345 Content-Length: 310 v=0 o=- 8 2 IN IP4 192.168.1.2 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.2 t=0 0 m=audio 15654 RTP/AVP 107 0 8 18 101 a=alt:1 1 : njwQZFqS t0MkMBW6 192.168.1.2 15654 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (12 headers 13 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 Sending to 192.168.1.2 : 61424 (no NAT) Using INVITE request as basis request - NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. Found peer '501' for '501' from 192.168.1.2:61424 MyPBX*CLI> <--- Reliably Transmitting (no NAT) to 192.168.1.2:61424 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-66444c0eb6773214-1---d8754z-;received=192.168.1.2;rport=61424 From: "501"<sip:501@192.168.1.3>;tag=3d303e4e To: "81079030016641"<sip:81079030016641@192.168.1.3>;tag=as56a5c336 Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 1 INVITE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ab8c54e" Content-Length: 0 Scheduling destruction of SIP dialog 'NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA.' in 6720 ms (Method: INVITE) MyPBX*CLI> <--- SIP read from UDP:192.168.1.2:61424 ---> ACK sip:81079030016641@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-66444c0eb6773214-1---d8754z-;rport To: "81079030016641"<sip:81079030016641@192.168.1.3>;tag=as56a5c336 From: "501"<sip:501@192.168.1.3>;tag=3d303e4e Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- MyPBX*CLI> <--- SIP read from UDP:192.168.1.2:61424 ---> INVITE sip:81079030016641@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-6778c40f3135d725-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:501@192.168.1.2:61424> To: "81079030016641"<sip:81079030016641@192.168.1.3> From: "501"<sip:501@192.168.1.3>;tag=3d303e4e Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102u stamp 52345 Authorization: Digest username="501",realm="asterisk",nonce="0ab8c54e",uri="sip:81079030016641@192.168.1.3",response="3e4b3018e8ef65f59c4ef53a5d44468a",algorithm=MD5 Content-Length: 310 v=0 o=- 8 2 IN IP4 192.168.1.2 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.2 t=0 0 m=audio 15654 RTP/AVP 107 0 8 18 101 a=alt:1 1 : njwQZFqS t0MkMBW6 192.168.1.2 15654 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (13 headers 13 lines) --- Sending to 192.168.1.2 : 61424 (no NAT) Using INVITE request as basis request - NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. Found peer '501' for '501' from 192.168.1.2:61424 Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format BV32 for ID 107 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x2c000e (gsm|ulaw|alaw|h261|h263|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.2:15654 Peer doesn't provide video Looking for 81079030016641 in DLPN_DialPlan501 (domain 192.168.1.3) list_route: hop: <sip:501@192.168.1.2:61424> MyPBX*CLI> <--- Transmitting (no NAT) to 192.168.1.2:61424 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-6778c40f3135d725-1---d8754z-;received=192.168.1.2;rport=61424 From: "501"<sip:501@192.168.1.3>;tag=3d303e4e To: "81079030016641"<sip:81079030016641@192.168.1.3> Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 2 INVITE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:81079030016641@192.168.1.3> Content-Length: 0 <------------> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 [2012-05-06 21:13:42] WARNING[3117]: chan_sip.c:5489 create_addr: No such host: trunk-baza Really destroying SIP dialog '4f9cbecf58307c9964048fb860323172@127.0.0.1' Method: INVITE [2012-05-06 21:13:42] WARNING[3117]: app_dial.c:1762 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) Audio is at 192.168.1.3 port 13376 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP MyPBX*CLI> <--- Reliably Transmitting (no NAT) to 192.168.1.2:61424 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-6778c40f3135d725-1---d8754z-;received=192.168.1.2;rport=61424 From: "501"<sip:501@192.168.1.3>;tag=3d303e4e To: "81079030016641"<sip:81079030016641@192.168.1.3>;tag=as4cc4dc22 Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 2 INVITE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:81079030016641@192.168.1.3> Content-Type: application/sdp Content-Length: 260 v=0 o=root 512797649 512797649 IN IP4 192.168.1.3 s=Asterisk PBX SVN--r1826M c=IN IP4 192.168.1.3 t=0 0 m=audio 13376 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> MyPBX*CLI> <--- SIP read from UDP:192.168.1.2:61424 ---> ACK sip:81079030016641@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-ae1d5b27ae00a258-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:501@192.168.1.2:61424> To: "81079030016641"<sip:81079030016641@192.168.1.3>;tag=as4cc4dc22 From: "501"<sip:501@192.168.1.3>;tag=3d303e4e Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 2 ACK User-Agent: eyeBeam release 1102u stamp 52345 Authorization: Digest username="501",realm="asterisk",nonce="0ab8c54e",uri="sip:81079030016641@192.168.1.3",response="3e4b3018e8ef65f59c4ef53a5d44468a",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- MyPBX*CLI> <--- SIP read from UDP:192.168.1.2:61424 ---> BYE sip:81079030016641@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-740a7b4c0764c710-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:501@192.168.1.2:61424> To: "81079030016641"<sip:81079030016641@192.168.1.3>;tag=as4cc4dc22 From: "501"<sip:501@192.168.1.3>;tag=3d303e4e Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 3 BYE User-Agent: eyeBeam release 1102u stamp 52345 Authorization: Digest username="501",realm="asterisk",nonce="0ab8c54e",uri="sip:81079030016641@192.168.1.3",response="7a3d380454b1e01b6a65369d2cd07377",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.1.2 : 61424 (no NAT) MyPBX*CLI> <--- Transmitting (no NAT) to 192.168.1.2:61424 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:61424;branch=z9hG4bK-d8754z-740a7b4c0764c710-1---d8754z-;received=192.168.1.2;rport=61424 From: "501"<sip:501@192.168.1.3>;tag=3d303e4e To: "81079030016641"<sip:81079030016641@192.168.1.3>;tag=as4cc4dc22 Call-ID: NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA. CSeq: 3 BYE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 MyPBX*CLI> <------------> == Spawn extension (macro-trunkdial-failover-0.3, 9-CHANUNAVAIL, 1) exited non-zero on 'SIP/501-00000008' in macro 'trunkdial-failover-0.3' == Spawn extension (DLPN_DialPlan501, 81079030016641, 1) exited non-zero on 'SIP/501-00000008' Really destroying SIP dialog 'NTY5ZjVmNDJiMjc1YzEyNTNhY2Q0MzdiNTU1ZTVlZDA.' Method: BYE MyPBX*CLI> sip set debug of <--- SIP read from UDP:192.168.1.2:61424 ---> <-------------> MyPBX*CLI> sip set debug off SIP Debugging Disabled [2012-05-06 21:13:54] NOTICE[432]: utils.c:305 HostPoolUpdater: ======host:trunk-baza update dns====== MyPBX*CLI> exit [May 6 21:13:55] Executing last minute cleanups Asterisk ending (0). root:~>

Ничего необычного я в неём не обнаружил чтобы могла привести к такому результату. Вот привожу вывод еще пару комманд:

MyPBX*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 501/501 192.168.1.2 D 61424 OK (104 ms) MyPBX*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time qwerty.cnt.ru:5060 N 84995024657@ 285 Registered Sun, 06 May 2012 21:30:26 1 SIP registrations.

Подскажите пожалуйста что может вызывать такую ошибку? Если софтфон подключить непосредственно на VOIP сервер минуя астерикс - то звонок проходит замечательно.

Answer 1

1

сделайте вот так

core set verbose 3

выключите сип дебаг, он тут не нужен. и смотрите что происходит

han_sip.c:5489 create_addr: No such host: trunk-baza

это значит что у вас выполняется вызов вида SIP/trunk-baza а такого пира/транка не существует.

вообще я бы не рекомендовал приобретать такие поделки

ссылка удалить спам редактировать

ответил 2012-05-06 21:47:44 +0400

meral

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http://pro-sip.net/

Comments

Ну не знаю чем плоха такая поделка которая представляет собой маленькую коробочку с большим функционалом (который правда почему то не работает). Если скажите где взять аналогичный товар буду рад выслушать.

RainMan ( 2012-05-06 22:07:27 +0400 )редактировать

ну вы же сами ответили;) ибо не работает и за поддержкой вы обращаетесь не к ним, а к другим источникам. да и информации нет о ихнем вебе никакой по большому счету. используйте freepbx и соответсвенно коробочки с freepbx/trixbox embeded.

meral ( 2012-05-06 22:08:59 +0400 )редактировать

Answer 2

Спасибо за ответ.

Я сделал по вашей рекомендации

core set verbose 3

Понятного стало еще меньше в выводе информации после указанных вами рекомендаций, поэтому выкладываю без комментариев.

MyPBX*CLI> core set verbose 3 Verbosity was 2 and is now 3 [2012-05-06 21:58:57] NOTICE[432]: utils.c:305 HostPoolUpdater: ======host:trunk-baza update dns====== == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 -- Executing [81079030016641@DLPN_DialPlan501:1] Macro("SIP/501-0000000f", "trunkdial-failover-0.3,1,0,79030016641,trunk-baza,") in new stack -- Executing [s@macro-trunkdial-failover-0.3:1] NoOp("SIP/501-0000000f", "do call out") in new stack -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/501-0000000f", "0?Blacklist-Handle,s,1") in new stack -- Executing [s@macro-trunkdial-failover-0.3:3] GotoIf("SIP/501-0000000f", "0?4:5") in new stack -- Goto (macro-trunkdial-failover-0.3,s,5) -- Executing [s@macro-trunkdial-failover-0.3:5] Set("SIP/501-0000000f", "TCOUNT=4") in new stack -- Executing [s@macro-trunkdial-failover-0.3:6] Set("SIP/501-0000000f", "CDR(userfield)=Outbound") in new stack -- Executing [s@macro-trunkdial-failover-0.3:7] Set("SIP/501-0000000f", "OldCallerID=501") in new stack -- Executing [s@macro-trunkdial-failover-0.3:8] Set("SIP/501-0000000f", "TOUCH_MONITOR=501-79030016641") in new stack -- Executing [s@macro-trunkdial-failover-0.3:9] NoOp("SIP/501-0000000f", "") in new stack -- Executing [s@macro-trunkdial-failover-0.3:10] Set("SIP/501-0000000f", "TIMEOUT(absolute)=6000") in new stack Channel will hangup at 2012-05-06 23:39:05.972 ???. -- Executing [s@macro-trunkdial-failover-0.3:11] Set("SIP/501-0000000f", "DLSTAT=UNKNOW}") in new stack -- Executing [s@macro-trunkdial-failover-0.3:12] SetCktCustom("SIP/501-0000000f", "sendrpid,no,no") in new stack -- Executing [s@macro-trunkdial-failover-0.3:13] GotoIf("SIP/501-0000000f", "1>0?1-dial,1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing [1-dial@macro-trunkdial-failover-0.3:1] GotoIf("SIP/501-0000000f", "0?UNKNOW},1") in new stack -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/501-0000000f", "0?setdod,1:1-dial,3") in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,3) -- Executing [1-dial@macro-trunkdial-failover-0.3:3] Set("SIP/501-0000000f", "CALLERID(name)=84995024657") in new stack -- Executing [1-dial@macro-trunkdial-failover-0.3:4] Set("SIP/501-0000000f", "CALLERID(num)=84995024657") in new stack -- Executing [1-dial@macro-trunkdial-failover-0.3:5] Set("SIP/501-0000000f", "_SIPSRTP=") in new stack -- Executing [1-dial@macro-trunkdial-failover-0.3:6] Set("SIP/501-0000000f", "OUTDIALOPT=tTkKWwXx") in new stack -- Executing [1-dial@macro-trunkdial-failover-0.3:7] Dial("SIP/501-0000000f", "SIP/trunk-baza/079030016641,,tTkKWwXx") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 [2012-05-06 21:59:05] WARNING[3526]: chan_sip.c:5489 create_addr: No such host: trunk-baza [2012-05-06 21:59:05] WARNING[3526]: app_dial.c:1762 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [1-dial@macro-trunkdial-failover-0.3:8] Goto("SIP/501-0000000f", "1-CHANUNAVAIL,1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-CHANUNAVAIL,1) -- Executing [1-CHANUNAVAIL@macro-trunkdial-failover-0.3:1] Set("SIP/501-0000000f", "DLSTAT=9-CHANUNAVAIL") in new stack -- Executing [1-CHANUNAVAIL@macro-trunkdial-failover-0.3:2] Goto("SIP/501-0000000f", "2-dial,1)") in new stack -- Goto (macro-trunkdial-failover-0.3,2-dial,1) -- Executing [2-dial@macro-trunkdial-failover-0.3:1] Set("SIP/501-0000000f", "TCOUNT=5") in new stack -- Executing [2-dial@macro-trunkdial-failover-0.3:2] Goto("SIP/501-0000000f", "1-dial,1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing [1-dial@macro-trunkdial-failover-0.3:1] GotoIf("SIP/501-0000000f", "1?9-CHANUNAVAIL,1") in new stack -- Goto (macro-trunkdial-failover-0.3,9-CHANUNAVAIL,1) -- Executing [9-CHANUNAVAIL@macro-trunkdial-failover-0.3:1] Playback("SIP/501-0000000f", "all-busy-now-try-call-later") in new stack -- <SIP/501-0000000f> Playing 'all-busy-now-try-call-later.gsm' (language 'ru') == Spawn extension (macro-trunkdial-failover-0.3, 9-CHANUNAVAIL, 1) exited non-zero on 'SIP/501-0000000f' in macro 'trunkdial-failover-0.3' == Spawn extension (DLPN_DialPlan501, 81079030016641, 1) exited non-zero on 'SIP/501-0000000f' -- Executing [h@DLPN_DialPlan501:1] NoOp("SIP/501-0000000f", "no thing to do") in new stack MyPBX*CLI> core set verbose 2 Verbosity was 3 and is now 2 [2012-05-06 21:59:15] NOTICE[432]: utils.c:305 HostPoolUpdater: ======host:qwerty.cnt.ru update dns====== [2012-05-06 21:59:23] NOTICE[432]: utils.c:305 HostPoolUpdater: ======host:MyPBX update dns====== MyPBX*CLI> exit [May 6 21:59:27] Executing last minute cleanups Asterisk ending (0). root:~>

ну и? вот ваша строчка SIP/trunk-baza/079030016641,,tTkKWwXx. транка trunk-baza у вас нет в конфиге. тоесть либо создать транк с атким именем, либо разобраться что конкретно в вашей коробочке недоставили.
meral ( 2012-05-06 22:11:43 +0400 )редактировать

Answer 3

В комментариях нет форматирования кодов - продолжу диалог тут.

Сделал вот так:

root:/etc/asterisk> grep -r "trunk-baza" * extensions.conf:trunk-baza = SIP/trunk-baza extensions.conf:CID_trunk-baza = 84995024657 extensions.conf:exten = _810.,1,Macro(trunkdial-failover-0.3,1,0,${EXTEN:3},trunk-baza,) extensions.conf:exten = baza,hint,SIP/trunk-baza users.conf:[trunk-baza]

А вот кусок из users.conf

[trunk-baza] trunkname = baza displayhost = qwerty.cnt.ru port = 5060 fromdomain = qwerty.cnt.ru username = 84995024657 ysauth = 84995024657 secret = xxx enableserver = yes host = disallow = all allow = alaw,ulaw,gsm contact = 84995024657 context = from-trunk hasiax = no registeriax = no hassip = yes registersip = no hasexten = no call-limit = 256 trunkstyle = voip insecure = port,invite srtpcapable = no transport = udp fromuser = 84995024657 kr_diversion =

Значит trank-baza присутствует в конфигах.

ну так ребутните ваш девайс.
meral ( 2012-05-07 00:46:17 +0400 )редактировать

Answer 4

0

Сталкивался с такой проблемой. Решил опытным путем - добавить VOIP-канал sip-sip на 127.0.0.1 Так и не понял, с чем это связано - но вот заработало.

ссылка удалить спам редактировать

ответил 2012-09-27 10:12:29 +0400

poruchik
1 ● 1

обновил 2012-09-27 10:17:18 +0400

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А на голове перед коробочкой постоять не забыли?

zzuz ( 2012-09-27 12:07:29 +0400 )редактировать

MyPBX - create_addr: No such host

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