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@HonesTks
asterisk

Почему Asterisk не звонит на городские номера?

Добрый день. Использую freepbx Distro + АТС Alcatel.
Ситуация следующая, в которой никак не могу разобраться.
Завел трак и связал АТС с Астериском, пробовал позвонить на городской с сип-софтфона - все ок, на внутренние номера (они заведены в АТС) - тоже все ок.
При создании .call файла где я вписываю городской телефон, ругается:
[2015-02-05 21:27:13] WARNING[1721]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/244322246323838.call: Operation not permitted
    -- Attempting call on SIP/Trunk_Alcatel/89004267000 for application Playback(demo-congrats) (Retry 1)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Got SIP response 502 "Bad Gateway" back from 10.10.10.10:5060
       > Channel SIP/Trunk_Alcatel-00000009 was never answered.
[2015-02-05 21:27:14] NOTICE[3492]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)
[2015-02-05 21:27:14] NOTICE[3492]: pbx_spool.c:392 attempt_thread: Queued call to SIP/Trunk_Alcatel/89004267000 expired without completion after 0 attempts

Если вместо мобильного телефона вписать внутренний то звонок идет.
Скажите пожалуйста, что я делаю не так?
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@rostel
WARNING[1721]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/244322246323838.call: Operation not permitted

перед перемещением колфайла в /var/spool/asterisk/outgoing выставите ему соответствующие права 

*CLI> core set verbose 3
*CLI> sip set debug peer Trunk_Alcatel
перемещаете колфайл
выхлоп в студию
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15 комментариев
@HonesTks Автор вопроса 
Script started on Fri 06 Feb 2015 07:58:08 PM VLAT
]0;root@localhost:~[?1034h[root@localhost ~]# rebootstartservice sshd restart[11P./call.shmc[K
[K[root@localhost ~]# mcasterisk -vvvvvr
Asterisk 11.14.1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.14.1 currently running on localhost (pid = 1692)
localhost*CLI> core set verbose 3[16Gsip set debug peer Trunk_Alcatel[16Gcore set verbose 3[K

localhost*CLI> 
[0KConsole verbose was 5 and is now 3.

[Klocalhost*CLI> core set verbose 3[16Gsip set debug peer Trunk_Alcatel

localhost*CLI> 
[0KSIP Debugging Enabled for IP: 10.65.9.4

[Klocalhost*CLI> 
[0K[2015-02-06 19:59:43] [1;31mWARNING[0m[1741]: [1;37mpbx_spool.c[0m:[1;37m309[0m [1;37msafe_append[0m: Unable to set utime on /var/spool/asterisk/outgoing/23486156504648.call: Operation not permitted

[Klocalhost*CLI> 
[0K    -- Attempting call on SIP/Trunk_Alcatel/89004267333 for application Playback(demo-congrats) (Retry 1)

[Klocalhost*CLI> 
[0K  == Using SIP RTP TOS bits 184

[Klocalhost*CLI> 
[0K  == Using SIP RTP CoS mark 5

[Klocalhost*CLI> 
[0KAudio is at 17606

[Klocalhost*CLI> 
[0KAdding codec 100004 (alaw) to SDP

[Klocalhost*CLI> 
[0KAdding codec 100003 (ulaw) to SDP

[Klocalhost*CLI> 
[0KAdding non-codec 0x1 (telephone-event) to SDP

[Klocalhost*CLI> 
[0KReliably Transmitting (no NAT) to 10.65.9.4:5060:
INVITE sip:89004267333@10.65.9.4 SIP/2.0

Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77

Max-Forwards: 70

From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978

To: <sip:89004267333@10.65.9.4>

Contact: <sip:Unknown@10.65.14.247:5060>

Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060

CSeq: 102 INVITE

User-Agent: FPBX-2.11.0(11.14.1)

Date: Fri, 06 Feb 2015 09:59:43 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 258



v=0

o=root 770304551 770304551 IN IP4 10.65.14.247

s=Asterisk PBX 11.14.1

c=IN IP4 10.65.14.247

t=0 0

m=audio 17606 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv


---

[Klocalhost*CLI> 
[0K
<--- SIP read from UDP:10.65.9.4:5060 --->
SIP/2.0 100 Trying
To: <sip:89004267333@10.65.9.4>
From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978
Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
Content-Length: 0

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

[Klocalhost*CLI> 
[0K
<--- SIP read from UDP:10.65.9.4:5060 --->
SIP/2.0 502 Bad Gateway
User-Agent: OmniPCX Enterprise R9.1 i1.605.29
To: <sip:89004267333@10.65.9.4>;tag=e1ba17177d1cb58cdb44a959dc91d967
From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978
Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 502 "Bad Gateway" back from 10.65.9.4:5060
Transmitting (no NAT) to 10.65.9.4:5060:
ACK sip:89004267333@10.65.9.4 SIP/2.0

Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77

Max-Forwards: 70

From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978

To: <sip:89004267333@10.65.9.4>;tag=e1ba17177d1cb58cdb44a959dc91d967

Contact: <sip:Unknown@10.65.14.247:5060>

Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060

CSeq: 102 ACK

User-Agent: FPBX-2.11.0(11.14.1)

Content-Length: 0




---

[Klocalhost*CLI> 
[0K[2015-02-06 19:59:44] [1;33mNOTICE[0m[2400]: [1;37mpbx_spool.c[0m:[1;37m389[0m [1;37mattempt_thread[0m: Call failed to go through, reason (8) Congestion (circuits busy)
[2015-02-06 19:59:44] [1;33mNOTICE[0m[2400]: [1;37mpbx_spool.c[0m:[1;37m392[0m [1;37mattempt_thread[0m: Queued call to SIP/Trunk_Alcatel/89004267333 expired without completion after 0 attempts

[Klocalhost*CLI> 
[0KReally destroying SIP dialog '203c4ba9352392994569cc684d18e03a@10.65.14.247:5060' Method: INVITE

[Klocalhost*CLI> 
[0KReliably Transmitting (no NAT) to 10.65.9.4:5060:
OPTIONS sip:10.65.9.4 SIP/2.0

Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK2690a3ec

Max-Forwards: 70

From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as0256152c

To: <sip:10.65.9.4>

Contact: <sip:Unknown@10.65.14.247:5060>

Call-ID: 632632145529579d507bcc6e4402f852@10.65.14.247:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-2.11.0(11.14.1)

Date: Fri, 06 Feb 2015 10:00:13 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

[Klocalhost*CLI> 
[0K
<--- SIP read from UDP:10.65.9.4:5060 --->
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Contact: sip:10.65.9.4
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.1 i1.605.29
To: <sip:10.65.9.4>;tag=39e64c72a49c140608a50df2ba47e693
From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as0256152c
Call-ID: 632632145529579d507bcc6e4402f852@10.65.14.247:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK2690a3ec
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '632632145529579d507bcc6e4402f852@10.65.14.247:5060' Method: OPTIONS

[Klocalhost*CLI> 
[0KReliably Transmitting (no NAT) to 10.65.9.4:5060:
OPTIONS sip:10.65.9.4 SIP/2.0

Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK15392c1c

Max-Forwards: 70

From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as41c48d9c

To: <sip:10.65.9.4>

Contact: <sip:Unknown@10.65.14.247:5060>

Call-ID: 37205e9e5ccb0b4405a661ef119a1290@10.65.14.247:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-2.11.0(11.14.1)

Date: Fri, 06 Feb 2015 10:01:13 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

[Klocalhost*CLI> 
[0K
<--- SIP read from UDP:10.65.9.4:5060 --->
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Contact: sip:10.65.9.4
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.1 i1.605.29
To: <sip:10.65.9.4>;tag=f79fdf1fa8fdbb563747edac2ba5dfcc
From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as41c48d9c
Call-ID: 37205e9e5ccb0b4405a661ef119a1290@10.65.14.247:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK15392c1c
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '37205e9e5ccb0b4405a661ef119a1290@10.65.14.247:5060' Method: OPTIONS

[Klocalhost*CLI> 
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
]0;root@localhost:~[root@localhost ~]# asterisk -vvvvvr[10Prebootstartservice sshd restart[11P./call.shmc[K./call.shservice sshd restart[13Prestart[1Pbootasterisk -vvvvvr[Kasterisk -vvvvvr[Kexit
exit

Script done on Fri 06 Feb 2015 08:01:47 PM VLAT
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