Проблема с Huawei E173

[sis]

Всем хорошего настроения!
Я понимаю что в интернете много чего написано почти на все вопросы можно найти ответ но...
Проблема вот в чем взял готовый релюз AsteriskNOW11 32-bit с сайта Digium установил модем Huawei E173 все прописал как пишут на многих сайтах(http://pbx.gal.cv.ua/chan-dongle)!
Радости не было придела когда вызов пришел на софт-фон на мой ПО эта уже хорошо но столкнулся с проблемой исходящих звонков уже вторую ночь не сплю...
Пожалуйста напишите решение или укажите куда смотреть книгу по Asteriks читаю...
Заранее спасибо!
Если подробная информацию нужна я напишу. Просто не знаю какая вам нужна будет информация....

[ded]

Смотрим на http://forum.asterisk.ru/viewforum.php?f=5
вверху картинки про решение проблем. Пошагово решайте свои траблы.

Просто не знаю какая вам нужна будет информация....

примерно такая: Привет! Была проблема с исходящими через Huawei E173, решил так-то!

[sis]

Вот лаги при звонке на Asterisk 11.7.0 (# asteriks -rvv)

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[2014-04-30 10:45:48] WARNING[23284][C-000000d8]: channel.c:1002 channel_indicate: [Dongle/MEGAFON-0100000006] Don't know how to indicate condition 22
[2014-04-30 10:45:48] WARNING[23284][C-000000d8]: channel.c:1002 channel_indicate: [Dongle/MEGAFON-0100000006] Don't know how to indicate condition 22

<--- SIP read from UDP:192.168.0.2:41580 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK3ca5da81
Contact: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>
To: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>;tag=c07d7864
From: "MEGAFON"<sip:+79032478614@192.168.0.10>;tag=as5d8a2c4d
Call-ID: 4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>
[2014-04-30 10:45:48] WARNING[23284][C-000000d8]: channel.c:1002 channel_indicate: [Dongle/MEGAFON-0100000006] Don't know how to indicate condition 33
Scheduling destruction of SIP dialog '4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060' in 7104 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.2:41580:
CANCEL sip:100@192.168.0.2:41580;rinstance=979008b95320d95e SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK3ca5da81
Max-Forwards: 70
From: "MEGAFON" <sip:+79032478614@192.168.0.10>;tag=as5d8a2c4d
To: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>
Call-ID: 4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.11.0(11.7.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060' in 7104 ms (Method: INVITE)
== Spawn extension (macro-dial-one, s, 43) exited non-zero on 'Dongle/MEGAFON-0100000006' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'Dongle/MEGAFON-0100000006' in macro 'exten-vm'
== Spawn extension (from-did-direct, 100, 2) exited non-zero on 'Dongle/MEGAFON-0100000006'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Dongle/MEGAFON-0100000006' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'Dongle/MEGAFON-0100000006'

<--- SIP read from UDP:192.168.0.2:41580 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK3ca5da81
Contact: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>
To: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>;tag=c07d7864
From: "MEGAFON"<sip:+79032478614@192.168.0.10>;tag=as5d8a2c4d
Call-ID: 4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060
CSeq: 102 CANCEL
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.2:41580 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK3ca5da81
To: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>;tag=c07d7864
From: "MEGAFON"<sip:+79032478614@192.168.0.10>;tag=as5d8a2c4d
Call-ID: 4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.2:41580:
ACK sip:100@192.168.0.2:41580;rinstance=979008b95320d95e SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK3ca5da81
Max-Forwards: 70
From: "MEGAFON" <sip:+79032478614@192.168.0.10>;tag=as5d8a2c4d
To: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>;tag=c07d7864
Contact: <sip:+79032478614@192.168.0.10:5060>
Call-ID: 4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.7.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060' in 7104 ms (Method: INVITE)
Really destroying SIP dialog '4cfabeef2c1f318225a18f066058205c@192.168.0.10:5060' Method: INVITE
Reliably Transmitting (no NAT) to 192.168.0.2:41580:
OPTIONS sip:100@192.168.0.2:41580;rinstance=979008b95320d95e SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK2ed85d55
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.10>;tag=as27e6689a
To: <sip:100@192.168.0.2:41580;rinstance=979008b95320d95e>
Contact: <sip:Unknown@192.168.0.10:5060>
Call-ID: 756b04e56121ae9755b961a46259ceb7@192.168.0.10:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.7.0)
Date: Wed, 30 Apr 2014 06:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

При по пытки позвонить на мой сотовый префикс "9"

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:192.168.0.2:41580 --->
INVITE sip:989032478614@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-ca4cb6238d0e964d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.0.2:41580>
To: "989032478614"<sip:989032478614@192.168.0.10>
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
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: 312

v=0
o=- 8 2 IN IP4 192.168.0.2
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.0.2
t=0 0
m=audio 37776 RTP/AVP 107 9 0 8 18 101
a=alt:1 1 : k8LGi4XE CJTli6J1 192.168.0.2 37776
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) ---
Sending to 192.168.0.2:41580 (NAT)
Sending to 192.168.0.2:41580 (NAT)
Using INVITE request as basis request - MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
Found peer '100' for '100' from 192.168.0.2:41580

<--- Reliably Transmitting (no NAT) to 192.168.0.2:41580 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-ca4cb6238d0e964d-1---d8754z-;received=192.168.0.2;rport=41580
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
To: "989032478614"<sip:989032478614@192.168.0.10>;tag=as382e7547
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f4c3aee"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.' in 6656 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.2:41580 --->
ACK sip:989032478614@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-ca4cb6238d0e964d-1---d8754z-;rport
To: "989032478614"<sip:989032478614@192.168.0.10>;tag=as382e7547
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.2:41580 --->
INVITE sip:989032478614@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-036b7842415d205c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.0.2:41580>
To: "989032478614"<sip:989032478614@192.168.0.10>
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
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="100",realm="asterisk",nonce="5f4c3aee",uri="sip:989032478614@192.168.0.10",response="e0bdf4842180c82ca7810c180b366354",algorithm=MD5
Content-Length: 312

v=0
o=- 8 2 IN IP4 192.168.0.2
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.0.2
t=0 0
m=audio 37776 RTP/AVP 107 9 0 8 18 101
a=alt:1 1 : k8LGi4XE CJTli6J1 192.168.0.2 37776
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 192.168.0.2:41580 (no NAT)
Using INVITE request as basis request - MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
Found peer '100' for '100' from 192.168.0.2:41580
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found unknown media 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 - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.2:37776
Looking for 989032478614 in from-internal (domain 192.168.0.10)
list_route: hop: <sip:100@192.168.0.2:41580>

<--- Transmitting (no NAT) to 192.168.0.2:41580 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-036b7842415d205c-1---d8754z-;received=192.168.0.2;rport=41580
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
To: "989032478614"<sip:989032478614@192.168.0.10>
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:989032478614@192.168.0.10:5060>
Content-Length: 0


<------------>
[2014-04-30 10:56:55] WARNING[23296][C-000000e0]: channel.c:5956 ast_request: No channel type registered for 'datacard'
[2014-04-30 10:56:55] WARNING[23296][C-000000e0]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'datacard' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
Audio is at 15514
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.0.2:41580 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-036b7842415d205c-1---d8754z-;received=192.168.0.2;rport=41580
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
To: "989032478614"<sip:989032478614@192.168.0.10>;tag=as1f4738a2
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:989032478614@192.168.0.10:5060>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1294331950 1294331950 IN IP4 192.168.0.10
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.0.10
t=0 0
m=audio 15514 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

<------------>

<--- SIP read from UDP:192.168.0.2:41580 --->
CANCEL sip:989032478614@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-036b7842415d205c-1---d8754z-;rport
To: "989032478614"<sip:989032478614@192.168.0.10>
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
CSeq: 2 CANCEL
User-Agent: eyeBeam release 1102u stamp 52345
Authorization: Digest username="100",realm="asterisk",nonce="5f4c3aee",uri="sip:989032478614@192.168.0.10",response="df735e6da8f8e7a198670320bd15e660",algorithm=MD5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.0.2:41580 (no NAT)

<--- Reliably Transmitting (no NAT) to 192.168.0.2:41580 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-036b7842415d205c-1---d8754z-;received=192.168.0.2;rport=41580
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
To: "989032478614"<sip:989032478614@192.168.0.10>;tag=as1f4738a2
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.0.2:41580 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:41580;branch=z9hG4bK-d8754z-036b7842415d205c-1---d8754z-;received=192.168.0.2;rport=41580
From: "sis"<sip:100@192.168.0.10>;tag=3017721c
To: "989032478614"<sip:989032478614@192.168.0.10>;tag=as1f4738a2
Call-ID: MDU2YjNmOTFiYWEzNzU1NDgwNjI0NTY5ZWQ2ZTI3ZWU.
CSeq: 2 CANCEL
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/100-000000e3' in macro 'outisbusy'
== Spawn extension (from-internal, 989032478614, 7) exited non-zero on 'SIP/100-000000e3'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-000000e3'

Я так думаю что где та в # nano /etc/asterisk/extensions.conf не дописано или наоборот...

[Vlad1983]

читаем статью про chan_dongle, а пишем:

[2014-04-30 10:56:55] WARNING[23296][C-000000e0]: channel.c:5956 ast_request: No channel type registered for 'datacard'

[sis]

Ну не ставится он нормально вылетает ошибка:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[root@localhost asterisk-chan-dongle-asterisk11]# aclocal
configure.in:10: warning: AC_ARG_PROGRAM was called before AC_CANONICAL_TARGET
../../lib/autoconf/general.m4 AC_CANONICAL_TARGET is expanded from...
../../lib/autoconf/general.m4 AC_CANONICAL_SYSTEM is expanded from...
configure.in:10: the top level
[root@localhost asterisk-chan-dongle-asterisk11]# autoconf
configure.in:10: warning: AC_ARG_PROGRAM was called before AC_CANONICAL_TARGET
../../lib/autoconf/general.m4 AC_CANONICAL_TARGET is expanded from...
../../lib/autoconf/general.m4 AC_CANONICAL_SYSTEM is expanded from...
configure.in:10: the top level
[root@localhost asterisk-chan-dongle-asterisk11]# automake -a
configure.in:10: warning: AC_ARG_PROGRAM was called before AC_CANONICAL_TARGET
../../lib/autoconf/general.m4 AC_CANONICAL_TARGET is expanded from...
../../lib/autoconf/general.m4 AC_CANONICAL_SYSTEM is expanded from...
configure.in:10: the top level
configure.in:7: installing `./missing'
automake: no `Makefile.am' found for any configure output
[root@localhost asterisk-chan-dongle-asterisk11]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking target system type... i686-pc-linux-gnu
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for a thread-safe mkdir -p... /bin/mkdir -p
checking for gawk... gawk
checking whether make sets $(MAKE)... yes
checking target system type... (cached) i686-pc-linux-gnu
checking host system type... (cached) i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking for style of include used by make... GNU
checking dependency style of gcc... none
checking how to run the C preprocessor... gcc -E
checking for a BSD-compatible install... /usr/bin/install -c
checking for strip... strip
checking for rm... rm
checking for library containing iconv... none required
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking fcntl.h usability... yes
checking fcntl.h presence... yes
checking for fcntl.h... yes
checking for stdlib.h... (cached) yes
checking for string.h... (cached) yes
checking sys/time.h usability... yes
checking sys/time.h presence... yes
checking for sys/time.h... yes
checking termios.h usability... yes
checking termios.h presence... yes
checking for termios.h... yes
checking whether asterisk.h in ../include... no
checking whether asterisk.h in /usr/include... no
checking whether asterisk.h in /usr/local/include... no
checking whether asterisk.h in /opt/local/include... no
configure: error: Can't find "asterisk.h"
[root@localhost asterisk-chan-dongle-asterisk11]#

Как описано во многих статьях так и ставлю но... поэтому и был взят http://pbx.gal.cv.ua/chan-dongle готовый модуль...

Или на 11 мало кто собирает под свисток?

[Vlad1983]

Код: Выделить всё

yum install asterisk-devel

[sis]

Vlad1983 Спасибо все встало, но возвращаемся к моему второму сообщения все как было так и осталось...((((

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[root@localhost asterisk-chan-dongle-asterisk11]# make install
./config.status
config.status: creating Makefile
config.status: creating config.h
config.status: config.h is unchanged
config.status: executing depfiles commands
strip chan_dongle.so
/usr/bin/install -c -m 755 chan_dongle.so /usr/lib/asterisk/modules
[root@localhost asterisk-chan-dongle-asterisk11]#

# nano /etc/asterisk/dongle.conf

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[general]

interval=15 ; Number of seconds between trying to connect to devices

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; Dongle channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The Dongle channel can't accept jitter,
; thus an enabled jitterbuffer on the receive Dongle side will always
; be used if the sending side can create jitter.

;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a Dongle
; channel. Defaults to "no".

;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.

;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a Dongle
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

;jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
; The option represents the number of milliseconds by which the new jitter buffer
; will pad its size. the default is 40, so without modification, the new
; jitter buffer will set its size to the jitter value plus 40 milliseconds.
; increasing this value may help if your network normally has low jitter,
; but occasionally has spikes.

;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

[defaults]
; now you can set here any not required device settings as template
; sure you can overwrite in any [device] section this default values

context=from-pstn ; context for incoming calls
group=0 ; calling group
rxgain=0 ; increase the incoming volume; may be negative
txgain=0 ; increase the outgoint volume; may be negative
autodeletesms=yes ; auto delete incoming sms
resetdongle=yes ; reset dongle during initialization with ATZ command
u2diag=-1 ; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres=no ; use the caller ID presentation or not
callingpres=allowed_passed_screen ; set caller ID presentation by default use default network settings
disablesms=no ; disable of SMS reading from device when received
; chan_dongle has currently a bug with SMS reception. When a SMS gets in during a
; call chan_dongle might crash. Enable this option to disable sms reception.
; default = no

language=en ; set channel default language
smsaspdu=yes ; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
mindtmfgap=45 ; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration=80 ; minimal DTMF tone duration in ms
mindtmfinterval=200 ; minimal interval between ends of DTMF of same digits in ms

callwaiting=auto ; if 'yes' allow incoming calls waiting; by default use network settings
; if 'no' waiting calls just ignored
disable=no ; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section

initstate=start ; specified initial state of device, must be one of 'stop' 'start' 'remote'
; 'remove' same as 'disable=yes'

exten=+79255336559 ; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)

dtmf=relax ; control of incoming DTMF detection, possible values:
; off - off DTMF tones detection, voice data passed to asterisk unaltered
; use this value for gateways or if not use DTMF for AVR or inside dialplan
; inband - do DTMF tones detection
; relax - like inband but with relaxdtmf option
; default is 'relax' by compatibility reason

; dongle required settings
[MTS1]
audio=/dev/ttyUSB1 ; tty port for audio connection; no default value
data=/dev/ttyUSB2 ; tty port for AT commands; no default value
imei=352216045800436
;exten=79255336559
; if audio and data set together with imei and/or imsi audio and data has precedence
; you can use both imei and imsi together in this case exact match by imei and imsi required

# nano sip.conf не трогал как есть настроил через сайт добавил расширение(sip100) и настройки сети и нат...

Подредактирован # nano /etc/asterisk/extensions.conf
добавлены строки

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

exten => _7X.,1,Dial(Dongle/MTS1/holdother:+${FILTER(0-9,${EXTEN})})
exten => _+7X.,1,Dial(Dongle/MTS1/holdother:+${FILTER(0-9,${EXTEN})})
exten => _8X.,1,Dial(Dongle/MTS1/holdother:+7${FILTER(0-9,${EXTEN:1})})
exten => _007X.,1,Dial(Dongle/MTS1/holdother:+7${FILTER(0-9,${EXTEN:3})})
exten => h,1,Hangup()

[Vlad1983]

смотреть в extensions*.conf

в dongle.conf audio и data лучше не прописывать вместе с imei или imsi оно принципиально для разных методов поиска свистка

[sis]

# nano /etc/asterisk/extensions_additional.conf очень большой что имменно искать там?
# nano /etc/asterisk/extensions_custom.conf

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[from-gsm]
exten => s,1,Set(CALLERID(all)=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=8${CALLERID(num):2})
exten => s,n,goto(from-trunk,${IMEI},1)

# nano /etc/asterisk/extensions_custom.conf.sample

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

; This file contains example extensions_custom.conf entries.
; extensions_custom.conf should be used to include customizations
; to AMP's Asterisk dialplan.

; Extensions in AMP have access to the 'from-internal' context.
; The context 'from-internal-custom' is included in 'from-internal' by default

[from-internal-custom]
exten => 1234,1,Playback(demo-congrats) ; extensions can dial 1234
exten => 1234,2,Hangup()
exten => h,1,Hangup()
include => custom-recordme ; extensions can also dial 5678

; custom-count2four,s,1 can be used as a custom target for
; a Digital Receptionist menu or a Ring Group
[custom-count2four]
exten => s,1,SayDigits(1234)
exten => s,2,Hangup

; custom-recordme,5678,1 can be used as a custom target for
; a Digital Receptionist menu or a Ring Group
[custom-recordme]
exten => 5678,1,Wait(2)
exten => 5678,2,Record(/tmp/asterisk-recording:gsm)
exten => 5678,3,Wait(2)
exten => 5678,4,Playback(/tmp/asterisk-recording)
exten => 5678,5,Wait(2)
exten => 5678,6,Hangup

; This macro is for dev purposes and just dumps channel/app variables. Useful when designing new contexts.
[macro-dumpvars]
exten => s,1,Noop(ACCOUNTCODE=${ACCOUNTCODE})
exten => s,2,Noop(ANSWEREDTIME=${ANSWEREDTIME})
exten => s,3,Noop(BLINDTRANSFER=${BLINDTRANSFER})
exten => s,4,Noop(CALLERID=${CALLERID(all)})
exten => s,5,Noop(CALLERID(name)=${CALLERID(name)})
exten => s,6,Noop(CALLERID(number)=${CALLERID(number)})
exten => s,7,Noop(CALLINGPRES=${CALLINGPRES})
exten => s,8,Noop(CHANNEL=${CHANNEL})
exten => s,9,Noop(CONTEXT=${CONTEXT})
exten => s,10,Noop(DATETIME=${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
exten => s,11,Noop(DIALEDPEERNAME=${DIALEDPEERNAME})
exten => s,12,Noop(DIALEDPEERNUMBER=${DIALEDPEERNUMBER})
exten => s,13,Noop(DIALEDTIME=${DIALEDTIME})
exten => s,14,Noop(DIALSTATUS=${DIALSTATUS})
exten => s,15,Noop(DNID=${CALLERID(dnid)})
exten => s,16,Noop(EPOCH=${EPOCH})
exten => s,17,Noop(EXTEN=${EXTEN})
exten => s,18,Noop(HANGUPCAUSE=${HANGUPCAUSE})
exten => s,19,Noop(INVALID_EXTEN=${INVALID_EXTEN})
exten => s,20,Noop(LANGUAGE=${LANGUAGE()})
exten => s,21,Noop(MEETMESECS=${MEETMESECS})
exten => s,22,Noop(PRIORITY=${PRIORITY})
exten => s,23,Noop(RDNIS=${CALLERID(rdnis)}})
exten => s,24,Noop(SIPDOMAIN=${SIPDOMAIN})
exten => s,25,Noop(SIP_CODEC=${SIP_CODEC})
exten => s,26,Noop(SIPCALLID=${SIPCALLID})
exten => s,27,Noop(SIPUSERAGENT=${SIPUSERAGENT})
exten => s,29,Noop(TXTCIDNAME=${TXTCIDNAME})
exten => s,30,Noop(UNIQUEID=${UNIQUEID})
exten => s,31,Noop(TOUCH_MONITOR=${TOUCH_MONITOR})
exten => s,32,Noop(MACRO_CONTEXT=${MACRO_CONTEXT})
exten => s,33,Noop(MACRO_EXTEN=${MACRO_EXTEN})
exten => s,34,Noop(MACRO_PRIORITY=${MACRO_PRIORITY})

nano /etc/asterisk/extensions_override_freepbx.conf пустой

[Vlad1983]

для особо зрячих:

Название транка - MTS1, максимально каналов - 1, Специальный набор: Dongle/MTS1/$OUTNUM$

[2014-04-30 10:56:55] WARNING[23296][C-000000e0]: channel.c:5956 ast_request: No channel type registered for 'datacard'