Дебаг на Addpac
Received SIP PDU from ( 192.168.5.100:5182 )
OPTIONS sip:79298835533@192.168.5.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK38acdae6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.100:5182>;tag=as73d493c2
To: <sip:79298835533@192.168.5.110>
Contact: <sip:asterisk@192.168.5.100:5182>
Call-ID: 0cab9ef20d2b01d46b1eeae901b25c94@192.168.5.100:5182
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:asterisk@192.168.5.100:5182>;tag=as73d493c2
To: <sip:79298835533@192.168.5.110>
Call-ID: 0cab9ef20d2b01d46b1eeae901b25c94@192.168.5.100:5182
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:79640095533@192.168.5.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK39ee3f91
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.100:5182>;tag=as61c3b219
To: <sip:79640095533@192.168.5.110>
Contact: <sip:asterisk@192.168.5.100:5182>
Call-ID: 5fcfbaef466b8f0c55781c4f095366c7@192.168.5.100:5182
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:asterisk@192.168.5.100:5182>;tag=as61c3b219
To: <sip:79640095533@192.168.5.110>
Call-ID: 5fcfbaef466b8f0c55781c4f095366c7@192.168.5.100:5182
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0
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Вот дебаг со второй симки которая работает
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:79640095533@192.168.5.100:5182)
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:79640095533@192.168.5.100:5182 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112
From: <sip:79640095533@192.168.5.100>;tag=ec58d244a4
To: <sip:79640095533@192.168.5.100:5182>
Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
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:79640095533@192.168.5.110>
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:79679328252@192.168.5.100>;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:79640095533@192.168.5.100>;tag=ec58d244a4
To: <sip:79640095533@192.168.5.100:5182>
Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
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:79640095533@192.168.5.100:5182>
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:79640095533@192.168.5.100>;tag=ec58d244a4
To: <sip:79640095533@192.168.5.100:5182>;tag=as14eba75c
Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
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:79640095533@192.168.5.100:5182>
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:79640095533@192.168.5.100>;tag=ec58d244a4
To: <sip:79640095533@192.168.5.100:5182>;tag=as14eba75c
Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
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:79640095533@192.168.5.100:5182>
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:79640095533@192.168.5.100:5182 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112
From: <sip:79640095533@192.168.5.100>;tag=ec58d244a4
To: <sip:79640095533@192.168.5.100:5182>;tag=as14eba75c
Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
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:79640095533@192.168.5.100:5182 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
