Hello. I got some problem for Asterisk SIP trunk.
It’s Asterisk 11.17.1. SIP trunk from an operator.
Outgoing call : Everything is ok
Incoming call : Call signaling is ok but audio RTP is one way. Peer A,B and C can send audio packet to Peer X but it isn’t able to receive audio packet from Peer X.
Network :
[Asterisk trunk] ------- [VoIP Provider's Router] ---------- [VoIP Provider's SBC] --------- [Internet]----------[Peer X]
IP: 192.168.0.5 Local IP : 192.168.0.1 IP: 10.202.249.124
External IP : 10.228.238.238
│
│
[Peer A] -------------------------┘
IP: 192.168.0.95 │
│
[Peer B] -------------------------┘
IP: 192.168.0.xx │
│
[Peer C] -------------------------┘
IP: 192.168.0.XX
Peer A,B and C belong the same ring group for the incoming call.
[sip.conf]
nat = yes
localnet = 192.168.0.0/255.255.255.0
canreinvite = yes (Testing result is the same whichever I choose)
qualify = yes (Also same issue even though I chose NO)
The VoIP Provider’s Router is already done DMZ to Asterisk trunk(192.168.0.5) and opened all ports for RTP.
Additionally, I’ve tested that changed IP address Peer A to 192.168.0.5 and tried direct registration Peer A to SBC without SIP trunk. That’s working very well both outgoing/incoming. There’s no one-way audio issue.
Only it happened when I used Asterisk SIP trunk.
SDP log :
<--- SIP read from UDP:10.202.249.124:5060 --->
INVITE sip:028889266@192.168.0.5:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>
CSeq: 1 INVITE
Max-Forwards: 68
Contact: <sip:023514637@10.202.249.124:5060;transport=udp>
Min-SE: 90
Session-Expires: 1800
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel,timer
Content-Length: 330
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 31938512 31938512 IN IP4 10.202.249.124
s=Sip Call
c=IN IP4 10.202.249.124
t=0 0
m=audio 65488 RTP/AVP 8 0 18 4 2 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
--- (15 headers 14 lines) ---
0KSending to 10.202.249.124:5060 (no NAT)
Sending to 10.202.249.124:5060 (no NAT)
Using INVITE request as basis request - imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
0KFound peer 'trunk_1' for '023514637' from 10.202.249.124:5060
0K == Using SIP RTP CoS mark 5
0KFound RTP audio format 8
0KFound RTP audio format 0
0KFound RTP audio format 18
0KFound RTP audio format 4
0KFound RTP audio format 2
0KFound RTP audio format 96
0KFound audio description format PCMA for ID 8
0KFound audio description format PCMU for ID 0
0KFound audio description format G729 for ID 18
0KFound audio description format G723 for ID 4
0KFound audio description format G726-32 for ID 2
0KFound audio description format telephone-event for ID 96
0KCapabilities: us - (ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
0KPeer audio RTP is at port 10.202.249.124:65488
0KLooking for 028889266 in DID_trunk_1 (domain 192.168.0.5)
0Klist_route: hop: <sip:023514637@10.202.249.124:5060;transport=udp>
0K
<--- Transmitting (no NAT) to 10.202.249.124:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Length: 0
<------------>
0K -- Executing [028889266@DID_trunk_1:1] 1;36mGoto0m("1;35mSIP/trunk_1-000000d60m", "1;35mringroups-custom-10,s,10m") in new stack
0K -- Goto (ringroups-custom-10,s,1)
-- Executing [s@ringroups-custom-10:1] 1;36mNoOp0m("1;35mSIP/trunk_1-000000d60m", "1;35mOperator0m") in new stack
0K -- Executing [s@ringroups-custom-10:2] 1;36mDial0m("1;35mSIP/trunk_1-000000d60m", "1;35mSIP/1603&SIP/5900,60,i0m") in new stack
0K == Using SIP VIDEO CoS mark 6
0K == Using SIP RTP CoS mark 5
0K == Using SIP VIDEO CoS mark 6
0K == Using SIP RTP CoS mark 5
0KAudio is at 21032
0KVideo is at 192.168.0.5:23870
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200003 (h263p) to SDP
0KAdding video codec 200002 (h263) to SDP
0KAdding non-codec 0x1 (telephone-event) to SDP
0KReliably Transmitting (NAT) to 192.168.0.5:34465:
INVITE sip:1603@192.168.0.5:34465;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK7bdbb55b;rport
Max-Forwards: 70
From: <sip:023514637@192.168.0.5>;tag=as5ec9ca94
To: <sip:1603@192.168.0.5:34465;transport=udp>
Contact: <sip:023514637@192.168.0.5:5060>
Call-ID: 0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 07 Jan 2011 17:06:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 573
v=0
o=root 44622586 44622586 IN IP4 192.168.0.5
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.0.5
b=CT:2048
t=0 0
m=audio 21032 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 23870 RTP/AVP 99 98 34
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
0K
<--- SIP read from UDP:192.168.0.5:34465 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 192.168.0.5:5060;rport=5060;branch=z9hG4bK7bdbb55b
From: <sip:023514637@192.168.0.5>;tag=as5ec9ca94
To: <sip:1603@192.168.0.5:34465;transport=udp>
Call-ID: 0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
0K -- Called SIP/1603
0KAudio is at 28082
Video is at 192.168.0.5:21398
0KAdding codec 100003 (ulaw) to SDP
0KAdding codec 100002 (gsm) to SDP
0KAdding video codec 200004 (h264) to SDP
0KAdding video codec 200002 (h263) to SDP
0KAdding video codec 200003 (h263p) to SDP
0KAdding non-codec 0x1 (telephone-event) to SDP
0KReliably Transmitting (NAT) to 192.168.0.95:61843:
INVITE sip:5900@192.168.0.95:61843;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK7b9cb733;rport
Max-Forwards: 70
From: <sip:023514637@192.168.0.5>;tag=as35a93267
To: <sip:5900@192.168.0.95:61843;ob>
Contact: <sip:023514637@192.168.0.5:5060>
Call-ID: 4100bfde04ccb77f592c674548434365@192.168.0.5:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 07 Jan 2011 17:06:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 577
v=0
o=root 1551422348 1551422348 IN IP4 192.168.0.5
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.0.5
b=CT:2048
t=0 0
m=audio 28082 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 21398 RTP/AVP 99 34 98
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
---
0K -- Called SIP/5900
0K -- SIP/5900-000000d8 connected line has changed. Saving it until answer for SIP/trunk_1-000000d6
0K
<--- SIP read from UDP:192.168.0.5:34465 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.5:5060;rport=5060;branch=z9hG4bK7bdbb55b
From: <sip:023514637@192.168.0.5>;tag=as5ec9ca94
To: <sip:1603@192.168.0.5:34465;transport=udp>;tag=1523702966
Contact: <sip:1603@192.168.0.5:34465;transport=udp>
Call-ID: 0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
--- (9 headers 0 lines) ---
0Klist_route: hop: <sip:1603@192.168.0.5:34465;transport=udp>
0K -- SIP/1603-000000d7 connected line has changed. Saving it until answer for SIP/trunk_1-000000d6
0K -- SIP/1603-000000d7 is ringing
0K
<--- Transmitting (no NAT) to 10.202.249.124:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Length: 0
<------------>
0K
<--- SIP read from UDP:192.168.0.95:61843 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.5:5060;rport=5060;received=192.168.0.5;branch=z9hG4bK7b9cb733
Call-ID: 4100bfde04ccb77f592c674548434365@192.168.0.5:5060
From: <sip:023514637@192.168.0.5>;tag=as35a93267
To: <sip:5900@192.168.0.95;ob>
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
0K
<--- SIP read from UDP:192.168.0.95:61843 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.5:5060;rport=5060;received=192.168.0.5;branch=z9hG4bK7b9cb733
Call-ID: 4100bfde04ccb77f592c674548434365@192.168.0.5:5060
From: <sip:023514637@192.168.0.5>;tag=as35a93267
To: <sip:5900@192.168.0.95;ob>;tag=63c6b52e2d6e4347942eefa784af4246
CSeq: 102 INVITE
Contact: "5900" <sip:5900@192.168.0.95:61843;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:5900@192.168.0.95:61843;ob>
0K -- SIP/5900-000000d8 is ringing
0KReally destroying SIP dialog '799c349549ca218a3edf45f4752b2398@192.168.0.5:5060' Method: INVITE
0K
<--- SIP read from UDP:192.168.0.95:61843 --->
<------------->
0K
<--- SIP read from UDP:192.168.0.5:34465 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.5:5060;rport=5060;branch=z9hG4bK7bdbb55b
From: <sip:023514637@192.168.0.5>;tag=as5ec9ca94
To: <sip:1603@192.168.0.5:34465;transport=udp>;tag=1523702966
Contact: <sip:1603@192.168.0.5:34465;transport=udp>
Call-ID: 0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 860
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=doubango 1983 678902 IN IP4 192.168.0.5
s=-
c=IN IP4 192.168.0.5
t=0 0
m=audio 4466 RTP/AVP 0 101
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=sendrecv
a=ssrc:160639591 cname:e346b09f3bf0abd345ab01cdbb048d40
a=ssrc:160639591 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:160639591 label:doubango@audio
m=video 43894 RTP/AVP 98 34
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack
a=rtcp-fb:* goog-remb
a=rtcp-fb:* doubs-jcng
a=label:101
a=content:main
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=sendrecv
a=ssrc:327327727 cname:1ed807f435c5b488da35874d4ff23f31
a=ssrc:327327727 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:327327727 label:doubango@video
<------------->
--- (10 headers 32 lines) ---
0KFound RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 98
Found RTP video format 34
Found video description format H263-1998 for ID 98
Found video description format H263 for ID 34
0KCapabilities: us - (gsm|ulaw|h263|h263p|h264), peer - audio=(ulaw)/video=(h263|h263p)/text=(nothing), combined - (ulaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
0KPeer audio RTP is at port 192.168.0.5:4466
Peer video RTP is at port 192.168.0.5:43894
0Klist_route: hop: <sip:1603@192.168.0.5:34465;transport=udp>
0Kset_destination: Parsing <sip:1603@192.168.0.5:34465;transport=udp> for address/port to send to
set_destination: set destination to 192.168.0.5:34465
0KTransmitting (NAT) to 192.168.0.5:34465:
ACK sip:1603@192.168.0.5:34465;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK79b044bd;rport
Max-Forwards: 70
From: <sip:023514637@192.168.0.5>;tag=as5ec9ca94
To: <sip:1603@192.168.0.5:34465;transport=udp>;tag=1523702966
Contact: <sip:023514637@192.168.0.5:5060>
Call-ID: 0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
---
0K -- SIP/1603-000000d7 connected line has changed. Saving it until answer for SIP/trunk_1-000000d6
-- SIP/1603-000000d7 answered SIP/trunk_1-000000d6
0KScheduling destruction of SIP dialog '4100bfde04ccb77f592c674548434365@192.168.0.5:5060' in 32000 ms (Method: INVITE)
0KReliably Transmitting (NAT) to 192.168.0.95:61843:
CANCEL sip:5900@192.168.0.95:61843;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK7b9cb733;rport
Max-Forwards: 70
From: <sip:023514637@192.168.0.5>;tag=as35a93267
To: <sip:5900@192.168.0.95:61843;ob>
Call-ID: 4100bfde04ccb77f592c674548434365@192.168.0.5:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0
---
Scheduling destruction of SIP dialog '4100bfde04ccb77f592c674548434365@192.168.0.5:5060' in 32000 ms (Method: INVITE)
0KAudio is at 30448
0KAdding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
0K
<--- Reliably Transmitting (no NAT) to 10.202.249.124:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261
v=0
o=root 1929937670 1929937670 IN IP4 10.228.238.238
s=Asterisk PBX 11.17.1
c=IN IP4 10.228.238.238
t=0 0
m=audio 30448 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
<------------>
0K -- Locally bridging SIP/trunk_1-000000d6 and SIP/1603-000000d7
0K
<--- SIP read from UDP:192.168.0.95:61843 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.5:5060;rport=5060;received=192.168.0.5;branch=z9hG4bK7b9cb733
Call-ID: 4100bfde04ccb77f592c674548434365@192.168.0.5:5060
From: <sip:023514637@192.168.0.5>;tag=as35a93267
To: <sip:5900@192.168.0.95;ob>;tag=63c6b52e2d6e4347942eefa784af4246
CSeq: 102 CANCEL
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
0K
<--- SIP read from UDP:192.168.0.95:61843 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.5:5060;rport=5060;received=192.168.0.5;branch=z9hG4bK7b9cb733
Call-ID: 4100bfde04ccb77f592c674548434365@192.168.0.5:5060
From: <sip:023514637@192.168.0.5>;tag=as35a93267
To: <sip:5900@192.168.0.95;ob>;tag=63c6b52e2d6e4347942eefa784af4246
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
0KTransmitting (NAT) to 192.168.0.95:61843:
ACK sip:5900@192.168.0.95:61843;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK7b9cb733;rport
Max-Forwards: 70
From: <sip:023514637@192.168.0.5>;tag=as35a93267
To: <sip:5900@192.168.0.95:61843;ob>;tag=63c6b52e2d6e4347942eefa784af4246
Contact: <sip:023514637@192.168.0.5:5060>
Call-ID: 4100bfde04ccb77f592c674548434365@192.168.0.5:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
---
Scheduling destruction of SIP dialog '4100bfde04ccb77f592c674548434365@192.168.0.5:5060' in 32000 ms (Method: INVITE)
0K > 0x40b0c708 -- Probation passed - setting RTP source address to 192.168.0.5:4466
0KRetransmitting #1 (no NAT) to 10.202.249.124:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261
v=0
o=root 1929937670 1929937670 IN IP4 10.228.238.238
s=Asterisk PBX 11.17.1
c=IN IP4 10.228.238.238
t=0 0
m=audio 30448 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
---
0K > 0x40b13328 -- Probation passed - setting RTP source address to 192.168.0.5:43894
0KRetransmitting #2 (no NAT) to 10.202.249.124:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261
v=0
o=root 1929937670 1929937670 IN IP4 10.228.238.238
s=Asterisk PBX 11.17.1
c=IN IP4 10.228.238.238
t=0 0
m=audio 30448 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
---
0KRetransmitting #3 (no NAT) to 10.202.249.124:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261
v=0
o=root 1929937670 1929937670 IN IP4 10.228.238.238
s=Asterisk PBX 11.17.1
c=IN IP4 10.228.238.238
t=0 0
m=audio 30448 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
---
0KReally destroying SIP dialog '76033625-99ba-dacc-8dac-4907cc198382' Method: REGISTER
0K[Jan 7 17:07:06] 1;31mWARNING0m[26282]: 1;37mhttp.c0m:1;37m8870m 1;37mhttpd_helper_thread0m: Failed to set TCP_NODELAY on HTTP connection, getprotobyname("tcp") failed
0K[Jan 7 17:07:06] 1;31mWARNING0m[26282]: 1;37mhttp.c0m:1;37m8880m 1;37mhttpd_helper_thread0m: Some HTTP requests may be slow to respond.
0K
<--- SIP read from UDP:192.168.0.55:5060 --->
OPTIONS sip:192.168.0.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.55:5060;branch=z9hG4bK803517707179312532
From: "LINE 2" <sip:1201@192.168.0.5>;tag=243825834
To: <sip:192.168.0.5:5060>
Call-ID: 3416268521068-187331332230989@192.168.0.55
CSeq: 1 OPTIONS
Max-Forwards: 70
User-Agent: F52H/F52HP 2.5.348.66
Accept: application/sdp
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
0KSending to 192.168.0.55:5060 (no NAT)
0KLooking for s in public (domain 192.168.0.5)
0K
<--- Transmitting (no NAT) to 192.168.0.55:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.55:5060;branch=z9hG4bK803517707179312532;received=192.168.0.55
From: "LINE 2" <sip:1201@192.168.0.5>;tag=243825834
To: <sip:192.168.0.5:5060>;tag=as71e3fcbb
Call-ID: 3416268521068-187331332230989@192.168.0.55
CSeq: 1 OPTIONS
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.0.5:5060>
Accept: application/sdp
Content-Length: 0
<------------>
0KScheduling destruction of SIP dialog '3416268521068-187331332230989@192.168.0.55' in 32000 ms (Method: OPTIONS)
0KRetransmitting #4 (no NAT) to 10.202.249.124:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261
v=0
o=root 1929937670 1929937670 IN IP4 10.228.238.238
s=Asterisk PBX 11.17.1
c=IN IP4 10.228.238.238
t=0 0
m=audio 30448 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
0KRetransmitting #5 (no NAT) to 10.202.249.124:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261
v=0
o=root 1929937670 1929937670 IN IP4 10.228.238.238
s=Asterisk PBX 11.17.1
c=IN IP4 10.228.238.238
t=0 0
m=audio 30448 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
---
0K
<--- SIP read from UDP:192.168.0.5:34465 --->
BYE sip:023514637@192.168.0.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:34465;branch=z9hG4bK-992560726;rport
From: <sip:1603@192.168.0.5:34465>;tag=1523702966
To: <sip:023514637@192.168.0.5>;tag=as5ec9ca94
Call-ID: 0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060
CSeq: 1934992745 BYE
Content-Length: 0
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v1.0.10 (doubango r1328 - Vivex v1.0.1.3)
P-Preferred-Identity: <sip:1603@192.168.0.5>
<------------->
--- (18 headers 0 lines) ---
0KSending to 192.168.0.5:34465 (NAT)
0KScheduling destruction of SIP dialog '0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060' in 32000 ms (Method: BYE)
0K
<--- Transmitting (NAT) to 192.168.0.5:34465 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.5:34465;branch=z9hG4bK-992560726;received=192.168.0.5;rport=34465
From: <sip:1603@192.168.0.5:34465>;tag=1523702966
To: <sip:023514637@192.168.0.5>;tag=as5ec9ca94
Call-ID: 0eb17d7e587b9f08175ae4fd0c860895@192.168.0.5:5060
CSeq: 1934992745 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
0K == Spawn extension (ringroups-custom-10, s, 2) exited non-zero on 'SIP/trunk_1-000000d6'
0KScheduling destruction of SIP dialog 'imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000' in 32000 ms (Method: INVITE)
0K
<--- SIP read from UDP:192.168.0.95:61843 --->
<------------->
0KRetransmitting #6 (no NAT) to 10.202.249.124:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.249.124:5060;branch=z9hG4bK00jqnal241261mtu7b4rnq3dq3;received=10.202.249.124
From: <sip:023514637@10.202.249.124:5060;user=phone>;tag=ekl3gfdi-CC-29
To: <sip:028889266@192.168.0.5:5060;user=phone>;tag=as27d4229b
Call-ID: imfpdkgn3npsd2ki22oopncnomj3gn3e@SoftX3000
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028889266@10.228.238.238:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 261
v=0
o=root 1929937670 1929937670 IN IP4 10.228.238.238
s=Asterisk PBX 11.17.1
c=IN IP4 10.228.238.238
t=0 0
m=audio 30448 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
---
VoIP Provider’s Router Configuration :
event filter add vox all show mem
no reboot recovery-on-error
logging buffered size 16364
hostname TEST_IPPBx
interface FastEthernet 0/0
ip address 192.168.0.5 255.255.255.0
exit
interface FastEthernet 0/1
ip address 10.10.10.1 255.255.255.0
exit
interface dot11radio 0/0
dot11 qos wmm
exit
ip route 0.0.0.0 0.0.0.0 192.168.0.1
no snmp set-write-community private
no snmp set-read-community public
voice-default
voice-port 5/0
clock-source free_run
exit
voice-port 5/1
clock-source free_run
exit
voice-port 5/2
clock-source free_run
exit
voice-port 5/3
clock-source free_run
exit
voice-port 5/4
clock-source free_run
exit
voice-port 5/5
clock-source free_run
exit
voice-port 5/6
clock-source free_run
exit
voice-port 5/7
clock-source free_run
exit
voice-port 5/8
clock-source free_run
exit
dial-peer voice pots 0
insert-calling-number 028889266
pots-group 0
port 5/0
no shutdown
exit
dial-peer voice pots 1
insert-calling-number 029823331
pots-group 1
port 5/1
no shutdown
exit
dial-peer voice pots 2
insert-calling-number 022365909
pots-group 2
port 5/2
no shutdown
exit
dial-peer voice pots 3
insert-calling-number 0322635000
pots-group 3
port 5/3
no shutdown
exit
dial-peer voice pots 4
insert-calling-number 0322635001
pots-group 4
port 5/4
no shutdown
exit
dial-peer voice pots 5
insert-calling-number 0322635002
pots-group 5
port 5/5
no shutdown
exit
dial-peer voice pots 6
insert-calling-number 0322635003
pots-group 6
port 5/6
no shutdown
exit
dial-peer voice pots 7
insert-calling-number 0322635004
pots-group 7
port 5/7
no shutdown
exit
dial-peer voice voip 0
sig-protocol sip
dtmf-relay in-band
fax-relay passthrough
gw-ip-address 10.202.249.124
modem-passthrough
passthrough-mode reinvite
voip-coder-profile 0
sip-sdp-on-alert
sg3tog3 enable
no shutdown
exit
voice-routing
route 1
dial-peer pots-group 0
prefix-type outgoing called last
prefix 028889266 length 9
no loopback-routing
exit
route 2
dial-peer pots-group 1
prefix-type outgoing called last
prefix 029823331 length 9
no loopback-routing
exit
exit
sip-gateway
outbound-proxy 10.202.249.124
reg-dns-add 10.202.249.124
prox-dns-add 10.202.249.124
reg-ka 86400
gw-interface fastethernet 0/0
device-host-name 192.168.0.5
uri-contact ip-address
bye-on-refer
no shutdown
exit
voip-coder-profile 0
codec 0 g729ab 30
codec 1 g711a 30
codec 2 g711u 30
codec 3 g711u 10
exit
ibc-service
no auto-install
sip-proxy internal
shutdown
exit
end
Can I get some help? It makes me crazy.