Answer the question
In order to leave comments, you need to log in
D
D
Denis Prikhodko2015-02-05 14:49:48
Denis Prikhodko, 2015-02-05 14:49:48
Why doesn't Asterisk call landline numbers?
Good afternoon. I use freepbx Distro + PBX Alcatel.
The situation is the following, in which I can not figure it out.
I started the truck and connected the PBX with Asterisk, tried to call the landline from the sip softphone - everything is ok, to internal numbers (they are connected to the PBX) - everything is also ok.
When creating a .call file where I enter a landline phone, it swears:
[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 attemptsIf you enter an internal phone instead of a mobile phone, then the call goes.
Please tell me what am I doing wrong?
2 answer(s)
@rostel
before moving the colfile to /var/spool/asterisk/outgoing, set the appropriate rights for it
exhaust colfile to the studio
@HonesTks
V
Vladimir, 2015-02-05 @rostel
WARNING[1721]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/244322246323838.call: Operation not permitted
before moving the colfile to /var/spool/asterisk/outgoing, set the appropriate rights for it
*CLI> core set verbose 3
*CLI> sip set debug peer Trunk_Alcatelexhaust colfile to the studio
D
Denis Prikhodko, 2015-02-06 @HonesTks
Script started on Fri 06 Feb 2015 07:58:08 PM VLAT
]0;[email protected]:~[?1034h[[email protected] ~]# rebootstartservice sshd restart[11P./call.shmc[K
[K[[email protected] ~]# mcasterisk -vvvvvr
Asterisk 11.14.1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as404e1978
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>
From: "Unknown" <sip:[email protected]>;tag=as404e1978
Call-ID: [email protected]: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:[email protected]>;tag=e1ba17177d1cb58cdb44a959dc91d967
From: "Unknown" <sip:[email protected]>;tag=as404e1978
Call-ID: [email protected]: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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as404e1978
To: <sip:[email protected]>;tag=e1ba17177d1cb58cdb44a959dc91d967
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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 '[email protected]: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:[email protected]>;tag=as0256152c
To: <sip:10.65.9.4>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as0256152c
Call-ID: [email protected]: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 '[email protected]: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:[email protected]>;tag=as41c48d9c
To: <sip:10.65.9.4>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as41c48d9c
Call-ID: [email protected]: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 '[email protected]:5060' Method: OPTIONS
[Klocalhost*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
]0;[email protected]:~[[email protected] ~]# 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
Similar questions
F
E
T
M
Q
Didn't find what you were looking for?
Ask your questionAsk a Question
731 491 924 answers to any question