Dial failed for some reason with DIALSTATUS ...

[alex286]

Неизвестно по какой причине (возможно и мои шаловливые ручки), но в результате которой нет выхода на внешние транки.
Правда не все, а именно при наборе более 7 цифр.. Например: Всем известный Сипнет. При наборе "службы времени", либо "курса доллара" все работает.. Есть ответ, есть звук... В общем все нормально... а вот если больше цифер...то... ПРи этом красиво говорит, что все линии заняты...

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/Sipnet/74952253003
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/2032-000000c7", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/2032-000000c7", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/2032-000000c7", "RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/2032-000000c7", "34,1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("SIP/2032-000000c7", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/2032-000000c7", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/2032-000000c7", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/2032-000000c7", "CALLERID(number)=2032") in new stack
-- Executing [674952253003@from-internal:7] Macro("SIP/2032-000000c7", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/2032-000000c7", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/2032-000000c7", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/2032-000000c7", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/2032-000000c7", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/2032-000000c7> Playing 'all-circuits-busy-now.alaw' (language 'ru')
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- <SIP/2032-000000c7> Playing 'pls-try-call-later.alaw' (language 'ru')
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/2032-000000c7' in macro 'outisbusy'
== Spawn extension (from-internal, 674952253003, 7) exited non-zero on 'SIP/2032-000000c7'
-- Executing [h@from-internal:1] Macro("SIP/2032-000000c7", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/2032-000000c7", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/2032-000000c7", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/2032-000000c7", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("SIP/2032-000000c7", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("SIP/2032-000000c7", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,1
-- Executing [s@macro-hangupcall:18] NoOp("SIP/2032-000000c7", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("SIP/2032-000000c7", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp("SIP/2032-000000c7", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:26] GotoIf("SIP/2032-000000c7", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/2032-000000c7", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf("SIP/2032-000000c7", "1?theend") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup("SIP/2032-000000c7", "") in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/2032-000000c7' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2032-000000c7'
> doing dnsmgr_lookup for 'sipnet.ru'
192*CLI>

Интересует именно строка ""Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack "... вроде как "отфутболивание" оператором.. так?

[awsswa]

34 503 NORMAL_CIRCUIT_CONGESTION no circuit/channel available [Q.850] This cause indicates that there is no appropriate circuit/channel presently available to handle the call.

Нужен полный дебаг - тут не видно что они вам шлют

[alex286]

Пожалуйста... "выход" через "шестерку", которая отсекается

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

#

[zzuz]

SIP/2.0 401 Authentication required

Не смущает?

[ded]

Там после этого идёт авторизованный инвайт, с дайджестом, на который Тарио софтсвич отвечает уже
SIP/2.0 488 Not acceptable here

В общем тема из разряда "Не могу позвонить через Сипнет, помогите!"
http://asteriskforum.ru/viewtopic.php?t=789

[alex286]

Я больше не буду...

[alex286]

Там после этого идёт авторизованный инвайт, с дайджестом, на который Тарио софтсвич отвечает уже SIP/2.0 488 Not acceptable here

И кто виноват, что делать? Дороги или... Что ему надо отправить?

[ded]

Скорее всего у вас там на счету нет денег.

[alex286]

Да не с деньгами там все нормально... Было такое раньше, потом "пропало" и вот опять... SIP-настройки на всякий случай...

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: SipPhone
SDP Session Name: Asterisk PBX 1.8.11.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: Yes
Jitterbuffer forced: No
Jitterbuffer max size: 200
Jitterbuffer resync: 1000
Jitterbuffer impl: fixed
Jitterbuffer log: No

Network Settings:
---------------------------
SIP address remapping: Enabled using externaddr
Externhost: <none>
Externaddr: 89.235.177.214:0
Externrefresh: 60
Localnet: 192.168.10.0/255.255.255.0
192.168.10.0/255.255.255.0

Global Signalling Settings:
---------------------------
Codecs: 0xc000e (gsm|ulaw|alaw|h261|h263)
Codec Order: ulaw:20,gsm:20,alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: ru
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

----

[ded]

http://asteriskforum.ru/viewtopic.php?t=5471