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

23347 ● 24 ● 20 ● 177
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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