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sip trunk зарегистрированный

Reg.Time

sip.telecom.kz:5060 N XXXXXXXXX 705 Registered

Sun, 25 Aug 2013 15:33:40

1 SIP registrations.

Зарание прошу прошение за этот мусор, просто не чего не понимаю в логе
debag peers telecom
") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/102-00000025", "0?sub-pi ncheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/102-00000025", "0?disable trunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/102-00000025", "DIALNUMBER= 688777") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/102-00000025", "DIAL
TRUNKO PTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/102-00000025", "OUTBOUND
GRO UP=OUT2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/102-00000025", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/102-00000025", "0?skipout cid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/102-00000025", "DIAL
TRUNK_ OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/102-00000025", "outbound- callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/102-00000025", "0?Set (CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/102-00000025", "0?Set (REALCALLERIDNUM=102)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/102-00000025", "1?nor mcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/102-00000025", "USEROUTC ID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/102-00000025", "EMERGENC YCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/102-00000025", "TRUNKOUT CID=87172278265") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/102-00000025", "1?tru nkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/102-00000025", "1?Se t(CALLERID(all)=87172278265)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/102-00000025", "0?Se t(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/102-00000025", "0?Se t(CALLERID(all)=)") in new stack
-- Executing s@macro-outbound-callerid:15] ExecIf("SIP/102-00000025", "0?Se t(CALLERPRES()=prohibpassedscreen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/102-00000025", "1?sub-f lp-2,s,1") in new stack
-- Executing [s@sub-flp-2:1] ExecIf("SIP/102-00000025", "1?Return()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/102-00000025", "OUTNUM=6887 77") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/102-00000025", "custom=SIP/ siptelecom") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/102-00000025", "0?Set(DI ALTRUNKOPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/102-00000025", "dialout-t runk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/102-00000 025", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/102-00000025", "0?bypass ,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/102-00000025", "0?custom trunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/102-00000025", "SIP/siptel ecom/688777,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 19200
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 92.46.61.21:5060:
INVITE sip:688777@sip.telecom.kz:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.4:5060;branch=z9hG4bK463ed16f
Max-Forwards: 70
From: <sip:sip id@192.168.11.4="">;tag=as4f328245
To: <sip:688777@sip.telecom.kz:5060>
Contact: <sip:sip id@192.168.11.4:5060="">
Call-ID: 6be928143c58b3fa490cb51a36baa045@192.168.11.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(11.5.0)
Date: Sun, 25 Aug 2013 09:38:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 385896012 385896012 IN IP4 192.168.11.4
s=Asterisk PBX 11.5.0
c=IN IP4 192.168.11.4
t=0 0
m=audio 19200 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


-- Called SIP/siptelecom/688777</br>

<--- SIP read from UDP:92.46.61.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.4:5060;branch=z9hG4bK463ed16f

From: <sip:sip id@192.168.11.4:5060="">;tag=as4f328245
To: <sip:688777@sip.telecom.kz:5060>
Call-ID: 6be928143c58b3fa490cb51a36baa045@192.168.11.4:5060
CSeq: 102 INVITE

<-------------> --- (6 headers 0 lines) ---

<--- SIP read from UDP:92.46.61.21:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.11.4:5060;branch=z9hG4bK463ed16f
From: <sip:sip id@192.168.11.4:5060="">;tag=as4f328245
To: <sip:688777@sip.telecom.kz:5060>;tag=aprqngfrt-04p1mh20000c6
Call-ID: 6be928143c58b3fa490cb51a36baa045@192.168.11.4:5060
CSeq: 102 INVITE

<-------------> --- (6 headers 0 lines) ---
Transmitting (no NAT) to 92.46.61.21:5060:
ACK sip:688777@sip.telecom.kz:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.4:5060;branch=z9hG4bK463ed16f
Max-Forwards: 70
From: <sip:sip id@192.168.11.4="">;tag=as4f328245
To: <sip:688777@sip.telecom.kz:5060>;tag=aprqngfrt-04p1mh20000c6
Contact: <sip:380089925@192.168.11.4:5060>
Call-ID: 6be928143c58b3fa490cb51a36baa045@192.168.11.4:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(11.5.0)
Content-Length: 0


Scheduling destruction of SIP dialog '6be928143c58b3fa490cb51a36baa045@192.168.1 1.4:5060' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/102-00000025", "Dial faile d for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new st ack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/102-00000025", "s-CHANUNAV AIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/102-00000025", " RC=21") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/102-00000025", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/102-00000025", "continue,1 ") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/102-00000025", "1? noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/102-00000025", "TRUN K Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trun ks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/102-00000025", "CALLE RID(number)=102") in new stack
-- Executing [688777@from-internal:7] Macro("SIP/102-00000025", "outisbusy," ) in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/102-00000025", "") in new s tack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/102-00000025", "0?emergency,1 ") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/102-00000025", "0?intracompan y,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/102-00000025", "all-circuit s-busy-now&pls-try-call-later, noanswer") in new stack
-- <sip 102-00000025=""> Playing 'all-circuits-busy-now.gsm' (language 'ru')

0x2b42500a27e0 -- Probation passed - setting RTP source address to 192. 168.11.28:40038
-- <sip 102-00000025=""> Playing 'pls-try-call-later.gsm' (language 'ru') -- Executing [s@macro-outisbusy:5] Congestion("SIP/102-00000025", "20") in n ew stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/102-0000002 5' in macro 'outisbusy'
== Spawn extension (from-internal, 688777, 7) exited non-zero on 'SIP/102-0000 0025'
-- Executing [h@from-internal:1] Macro("SIP/102-00000025", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/102-00000025", "1?endmixmonc heck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/102-00000025", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/102-00000025", "1?nomeetmem on") in new stack
-- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("SIP/102-00000025", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/102-00000025", "1?noautomon ") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/102-00000025", "TOUCHMONITOR _OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/102-00000025", "1?noautomon 2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/102-00000025", "MONITOR
FILEN AME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/102-00000025", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/102-00000025", "1?skipblkvm ") in new stack
-- Goto (macro-hangupcall,s,48)
-- Executing [s@macro-hangupcall:48] GotoIf("SIP/102-00000025", "1?theend") in new stack
-- Goto (macro-hangupcall,s,50)
-- Executing [s@macro-hangupcall:50] AGI("SIP/102-00000025", "hangup.agi") i n new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <sip 102-00000025="">AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:51] Hangup("SIP/102-00000025", "") in new s tack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/102-00000 025' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/102-00000025'

trank setings
username=SIP/ID
secret=password
type=friend
nat=no
insecure=port,invite
fromuser=SIP/ID
dtmfmode=rfc2833
host=sip.telecom.kz
port=5060
disallow=all
canreinvite=no
allow=alaw&ulaw

Не судите строго!
Заранее ОГРОМНОЕ СПАСИБО!!

удалить закрыть спам изменить тег редактировать

спросил 2013-08-25 13:50:47 +0400

Jonik143 Gravatar Jonik143
1 1 1

обновил 2013-08-25 17:14:07 +0400

Comments

"это" не читается .

zzuz ( 2013-08-25 16:14:44 +0400 )редактировать

Да можно буду признателен, Можно еще решение проблемы Огромное спасибо

Jonik143 ( 2013-08-25 23:04:41 +0400 )редактировать

Спасибо хотя бы за одну запятую.

zzuz ( 2013-08-25 23:31:12 +0400 )редактировать

2 Ответа

0

Привет! У тебя получилось подключится к телекому? У меня при звонке говорит что номер занят.

ссылка удалить спам редактировать

ответил 2013-08-26 14:52:21 +0400

BRus Gravatar BRus
1 1

обновил 2013-08-26 16:57:01 +0400

Comments

Автор вопрос нормально задать не может. Вот если бы он fromdomain добавил бы в настройки пира , то может быть ему бы и повезло.

zzuz ( 2013-08-26 14:59:33 +0400 )редактировать
0
<--- SIP read from UDP:92.46.61.21:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.11.4:5060;branch=z9hG4bK463ed16f
From: <sip:sip id@192.168.11.4:5060="">;tag=as4f328245
To: <sip:688777@sip.telecom.kz:5060>;tag=aprqngfrt-04p1mh20000c6
Call-ID: 6be928143c58b3fa490cb51a36baa045@192.168.11.4:5060
CSeq: 102 INVITE

перевод нужен?

ссылка удалить спам редактировать

ответил 2013-08-25 17:30:43 +0400

meral Gravatar meral flag of Ukraine
21228 24 19 169
http://pro-sip.net/

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Задан: 2013-08-25 13:50:47 +0400

Просмотрен: 412 раз

Обновлен: Aug 26 '13

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.