Asterisk 13 + Addpac. Why is one of the GSM channels not working?

Question

Asterisk

Good evening everyone.
There is Asterisk + Addpac GS-1002 in the local network. A megaphone is placed in slot 1 on the Addpac, and a beeline is placed in slot 2.
There are no problems with the beeline, incoming and outgoing work properly. Trouble with the megaphone. Neither works. There are beeps, but the call to Asterisk does not fall.
sip.conf

[general]
trustrpid=yes
tcpenable=yes
useragent=D-link ; типа реальное железо. чтобы меньше привлекать внимание хацкеров которые ищут астериски
externip=192.168.5.100
localnet=192.168.5.0/24
qualify=yes               ; проверка доступности абонента - 2s - если больше, что считаем что недоступен
prematuremedia = no
progressinband = never
srvlookup=no
canreinvite=no              ; разрешает (yes) или запрещает (no) установку прямого соединения(минуя Asterisk).
directmedia=no              ; гнать трафик напрямую
allowguest = no             ; запрет регистрации "левых" аккунтов
transfer=yes                ; запрет трансфера вызовов глобально, включать вручную для нужных пиров
allowsubscribe=no           ; отказ от использования voicemail и соответствующего спама в консоли
alwaysauthreject=yes        ; на REGISTER Asterisk станет отвечать «401 Unathorized»
jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a SIP
jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize=300
jbimpl = adaptive           ; Jitterbuffer implementation, used on the receiving side of a SIP
jbresyncthreshold=100       ;adaptive jbimpl had troubles without thresh... Asterisk 1.6.2.9.
;jblog = yes                ; Enables jitterbuffer frame logging. Defaults to "no".
context=default             ; всех левых в дефротный контекст на отбой
relaxdtmf=yes
dtmfmode=auto
disallow=all
;allow=g729
;allow=g723
allow=alaw
allow=ulaw
allow=gsm
bindport=5182



[addpac_channels](!)            ; шаблон дублирующихся настроек для каналов шлюза
host=dynamic
;deny=0.0.0.0/0
permit=192.168.5.110
fromdomain=192.168.5.110
type=friend
context=from_trunk                ; входящие с SIP попадают в этот контекст в extensions.conf
qualify=yes
nat=no
canreinvite=no
insecure=port,invite            ; игнорировать порт и инвайт
disallow=all
allow=alaw
allow=ulaw
allow=gsm
maxcallbitrate=64
dtmfmode=rfc2833
port=5182


[79640095533](addpac_channels)
defaultuser=79640095533
secret=*******
call-limit=2
callerid=79640095533
relaxdtmf=yes

[79298835533](addpac_channels)
defaultuser=79298835533
secret=******
call-limit=1
callerid=79298835533
relaxdtmf=yes

[my_sip_user](!)
type=friend                   ; входящие и исходящие
Call-limit=2                  ; лимит количества одновременных звонков
host=dynamic                  ; обязательная регистрация
nat=force_rport,comedia       ; используется ли натирование адресов?
canreinvite=no                ; разрешает (yes) или запрещает (no) установку прямого соединения между участниками (минуя Asterisk).
directmedia=no                ; гнать трафик напрямую
dtmfmode=auto
disallow=all                  ; запретить все кодеки
;allow=alaw
;allow=ulaw
;allow=gsm                     ; разрешить нужные
;allow=g729
;allow=g723
port=5182
insecure=invite,port
context=_sip                  ; Контекст плана набора в extensions.conf, в который изначально попадают звонки с GSM-линий
transfer=yes
;transport=tcp                             ; В него включен основной контекст _sip для всего SIP-направления.

When sip show both peers register
Addpac config

!
! APOS(tm) configuration saved from vty
!  2016/11/06 19:41:52
!
version 8.51.011
!
hostname GS1002
!
username *****
!
!
script ntpdate default
 server ip time.nist.gov
 server ip time.windows.com
!
interface Loopback0
 ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
 ip address 192.168.5.110 255.255.255.0
 speed auto
 no qos-control
!
interface FastEthernet0/1
 ip address 192.168.10.1 255.255.255.0
 speed auto
 no qos-control
!
ip route 0.0.0.0 0.0.0.0 192.168.5.254
!
!
!
!
http server
!
logging command
logging event 4-warning
logging on
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
 protocol sip
 dtmf-relay rfc-2833
 fax protocol t38 redundancy 0
 fax rate 9600
 h323 call start fast
 h323 call tunnel enable
 no call-barring unconfigured-ip-address
 no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
 connection plar 79298835533
 ring detect-timeout 70
 dial-tone-generate
 caller-id enable
 caller-id type etsi
!
!
! GSM
voice-port 0/1
 connection plar 79640095533
 ring detect-timeout 70
 dial-tone-generate
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 0/2
 no caller-id enable
!
!
! FXO
voice-port 0/3
 no caller-id enable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
 destination-pattern 00T
 port 0/0
 call-waiting
 user-name 79298835533
 user-password ****
 translate-outgoing called-number 0
 diversion 1
!
dial-peer voice 1 pots
 destination-pattern 01T
 port 0/1
 call-waiting
 user-name 79640095533
 user-password ***
 translate-outgoing called-number 1
 preference 2
 diversion 2
!
!
!
! Voip peer configuration.
!
dial-peer voice 2000 voip
 destination-pattern T
 session target sip-server
 session protocol sip
 voice-class codec 1
 no vad
 dtmf-relay rtp-2833
 description asterisk
!
!
!
dial-peer call-hold h
dial-peer call-transfer h
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.5.100
 no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729
!
!
!
! Translation Rule configuration.
!
translation-rule 0
 rule 0      007T                     8T
!
translation-rule 1
 rule 0      017T                     8T
!
!
!
! SIP UA configuration.
!
sip-ua
 user-register
 sip-username addpac
 sip-password sip-secret
 sip-server 192.168.5.100 5182 1
 timeout treg 400
 called-party-number to-field
 remote-party-id
 session-refresh update
 register e164
!
!
! Tones
!
!
! SMS delivery configuration
!
sms-delivery
!
!
!
!
voip-interface ip FastEthernet0/0
!
line console
!
line vty
!
mobile dev-restart-by-unreg 180
mobile failed-call-retry 0
mobile ussd inter-frame-gap 100
mobile ussd balance-interval 120
mobile ussd retry-count 2
mobile ussd retry-interval 5
mobile ussd response-protection-time 5
mobile dev-restart-by-unknown-error
mobile cell-monitor 30
!
mobile 0/0
 gsm sms-language utf8
!
mobile 0/1
 gsm sms-language utf8
!

Where is the mistake?

Answer 1

enable debug on the gateway of everything and everything and watch

Answer 2

Have you tried using search?
www.awsswa.livejournal.com/22887.html

Answer 3

Debug on Addpac

Received SIP PDU from ( 192.168.5.100:5182 )
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK38acdae6
Max-Forwards: 70
From: "asterisk" <sip:[email protected]:5182>;tag=as73d493c2
To: <sip:[email protected]>
Contact: <sip:[email protected]:5182>
Call-ID: [email protected]:5182
CSeq: 102 OPTIONS
User-Agent: D-link
Date: Wed, 09 Nov 2016 22:39:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0



        Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK38acdae6
From: "asterisk" <sip:[email protected]:5182>;tag=as73d493c2
To: <sip:[email protected]>
Call-ID: [email protected]:5182
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0


        Received SIP PDU from ( 192.168.5.100:5182 )
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK39ee3f91
Max-Forwards: 70
From: "asterisk" <sip:[email protected]:5182>;tag=as61c3b219
To: <sip:[email protected]>
Contact: <sip:[email protected]:5182>
Call-ID: [email protected]:5182
CSeq: 102 OPTIONS
User-Agent: D-link
Date: Wed, 09 Nov 2016 22:39:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0



        Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK39ee3f91
From: "asterisk" <sip:[email protected]:5182>;tag=as61c3b219
To: <sip:[email protected]>
Call-ID: [email protected]:5182
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0
�

Here is the debug from the second sim card that works

135     <CEP    000100> : Call Received
136     <CEP    000100> : Call Received
137     <CEP    000100> : Call Initiated : calledNumber() crv(0) total(0)
138     <Call   179>    : ******  Call Created status(InitiatedByMobile) ver(8.5
139     <CEP    000100> : Decode CID : FFFFFF80  E 10  C 2B 37 39 36 37 39 33 32
140     <CEP    000100> : Mobile CID : time() callingNumber(79679328252) calling
141     <CEP    000100> : Calling number(79679328252)
142     <CEP    000100> : Call id(ecb02358-e738-ab8a-8143-0002a409fd2e) callNum(
143     <Call   179>    : MatchAllProcess After Sorted
                          <0>  id(2000) dest(T) prefer(0) selected(71)
144     <Call   179>    : Initiate callee with dial-peer(T) status(CalleeDetermi
145     <NetEP  179>    : InitiateOutCall: calledNum(79640095533) callingNum(796
146     <NetEP  179>    : DoCall: calledAddr(sip:[email protected]:5182)
147     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
148     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
149     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
150     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
151     <SIP    0>      : No authentication information available
152     <SIP    179>    : Send INVITE Request

        Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
INVITE sip:[email protected]:5182 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112
From: <sip:[email protected]>;tag=ec58d244a4
To: <sip:[email protected]:5182>
Call-ID: [email protected]
CSeq: 112 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Wed, 09 Nov 2016 23:27:40 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Contact: <sip:[email protected]>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 290
Max-Forwards: 70
Remote-Party-ID: <sip:[email protected]>;screen=yes;party=calling

v=0
o=79640095533 1478734060 1478734060 IN IP4 192.168.5.110
s=AddPac Gateway SDP
c=IN IP4 192.168.5.110
t=1478734060 0
m=audio 23358 RTP/AVP 8 0 18 101
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

        Received SIP PDU from ( 192.168.5.100:5182 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112;received=192.168
From: <sip:[email protected]>;tag=ec58d244a4
To: <sip:[email protected]:5182>
Call-ID: [email protected]
CSeq: 112 INVITE
Server: D-link
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5182>
Content-Length: 0


153     <SIP    179>    : Receive 100 Trying
154     <SIP    179>    : Transaction (112 INVITE) proceeding

        Received SIP PDU from ( 192.168.5.100:5182 )
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112;received=192.168
From: <sip:[email protected]>;tag=ec58d244a4
To: <sip:[email protected]:5182>;tag=as14eba75c
Call-ID: [email protected]
CSeq: 112 INVITE
Server: D-link
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5182>
Content-Type: application/sdp
Require: timer
Content-Length: 275

v=0
o=root 984719863 984719863 IN IP4 192.168.5.100
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.5.100
t=0 0
m=audio 6240 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=maxptime:150
a=sendrecv

155     <SIP    179>    : Receive 183 Session Progress
156     <SIP    179>    : Transaction (112 INVITE) proceeding
157     <SIP    179>    : Received Session Progress response
158     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
159     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
160     <SIP    179>    : Get SIP Audio MediaFormat : 8
161     <Call   179>    : PreConnected from(fffffffe)
162     <NetCon 179>    : Alert received (inband tone explicitly).
163     <Call   179>    : Alert from(fffffffe) pseudo(0) inband(1) status(Callee
164     <SIP    178>    : Set Terminated Success for 111 INVITE

        Received SIP PDU from ( 192.168.5.100:5182 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112;received=192.168
From: <sip:[email protected]>;tag=ec58d244a4
To: <sip:[email protected]:5182>;tag=as14eba75c
Call-ID: [email protected]
CSeq: 112 INVITE
Server: D-link
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5182>
Content-Type: application/sdp
Require: timer
Content-Length: 275

v=0
o=root 984719863 984719863 IN IP4 192.168.5.100
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.5.100
t=0 0
m=audio 6240 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=maxptime:150
a=sendrecv

165     <SIP    179>    : Receive 200 OK
166     <SIP    179>    : Received INVITE OK response
167     <SIP    179>    : Send ACK Request

        Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
ACK sip:[email protected]:5182 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112
From: <sip:[email protected]>;tag=ec58d244a4
To: <sip:[email protected]:5182>;tag=as14eba75c
Call-ID: [email protected]
CSeq: 112 ACK
Content-Length: 0
Max-Forwards: 70

168     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
169     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
170     <SIP    179>    : Get SIP Audio MediaFormat : 8
171     <Call   179>    : Connected from(fffffffe)
172     <NetEP  179>    : Call with sip:[email protected]:5182 establish
173     <SIP    179>    : Check Event Relation code(200)
174     <SIP    179>    : Set Terminated Success for 112 INVITE
175     <CEP    000100> : Disconnected(16) at Busy
176     <Call   179>    : Terminated from(100) this(Local:CallClear) before(NULL
177     <CEP    000100> : DisconnectCall at Idle
178     <SIP    179>    : ReleaseWithBYE
179     <SIP    179>    : Send BYE Request