I have configured FreePBX Phone System 300 (Asterisk 13.9) to use with a call center system.
The CC server is on LAN, so there is a simple SIP trunk to it.
However, the CC server sends two INVITE. First, with IN IP4 0.0.0.0 - and FreePBX, immetiately after receiving it, starts sending packets to 0.0.0.0:1234. Next INVITE is with IN IP4 192.168.100.250 (this is a valid address of the CC server) but the FreePBX doesn’t correct peer address and still sends packets to 0.0.0.0:1234. Canreinvite=yes in sip_general_additional.conf or in sip_additional.conf (in options of the trunk) doesn’t work.
An old PBX - Elastix 1.6 - has no problem with this behavior of the CC server.
Here a sample of session:
[2016-06-10 08:16:34] VERBOSE[2343] chan_sip.c:
<— SIP read from UDP:192.168.100.250:5060 —>
INVITE sip:600852586@192.168.100.200 SIP/2.0
From: "327843803"sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-26216-94f269f-38955b7d
Max-Forwards: 70
Supported: replaces
User-Agent: VocalcomOnnet
Contact: sip:192.168.100.250:5060
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY
Allow-Events: refer
Content-Type: application/sdp
Content-Length: 366
v=0
o=ONNET 1300002000 100082877 IN IP4 0.0.0.0
s=0_CALLMEDIA
i=ONNET
c=IN IP4 0.0.0.0
t=0 0
m=audio 1234 RTP/AVP 0 18 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=5.3; annexa=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
<------------->
[2016-06-10 08:16:34] VERBOSE[2343] chan_sip.c: — (14 headers 17 lines) —
[2016-06-10 08:16:34] VERBOSE[2343] chan_sip.c: Sending to 192.168.100.250:5060 (no NAT)
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Sending to 192.168.100.250:5060 (no NAT)
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Using INVITE request as basis request - 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found peer ‘asterisk-hermes’ for ‘192.168.100.250’ from 192.168.100.250:5060
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] netsock2.c: Using SIP RTP TOS bits 184
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] netsock2.c: Using SIP RTP CoS mark 5
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found RTP audio format 0
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found RTP audio format 18
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found RTP audio format 18
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found RTP audio format 4
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found RTP audio format 101
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found audio description format PCMU for ID 0
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found audio description format G729 for ID 18
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found audio description format G729 for ID 18
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found audio description format G723 for ID 4
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Found audio description format telephone-event for ID 101
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|g723|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Peer audio RTP is at port 0.0.0.0:1234
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c: Looking for 600852586 in from-internal (domain 192.168.100.200)
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] sip/route.c: sip_route_dump: route/path hop: sip:192.168.100.250:5060
[2016-06-10 08:16:34] VERBOSE[2343][C-00000004] chan_sip.c:
<— Transmitting (no NAT) to 192.168.100.250:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-26216-94f269f-38955b7d;received=192.168.100.250
From: "327843803"sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 1 INVITE
Server: FPBX-13.0.124(13.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:600852586@192.168.100.200:5060
Content-Length: 0
[2016-06-10 08:16:34] VERBOSE[3760][C-00000004] chan_sip.c:
<— Transmitting (no NAT) to 192.168.100.250:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-26216-94f269f-38955b7d;received=192.168.100.250
From: "327843803"sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 1 INVITE
Server: FPBX-13.0.124(13.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:600852586@192.168.100.200:5060
Content-Length: 0
<------------>
[2016-06-10 08:16:35] VERBOSE[2343] chan_sip.c: Really destroying SIP dialog ‘76589Y2UxZGM3M2MyZThmMmU0MmRhNWQxMTkxYmUxOTZiNWE’ Method: REGISTER
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] app_dial.c: DAHDI/i1/600852586-5 is making progress passing it to SIP/asterisk-hermes-00000004
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] chan_sip.c: Audio is at 17696
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] chan_sip.c: Adding codec ulaw to SDP
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] chan_sip.c:
<— Transmitting (no NAT) to 192.168.100.250:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-26216-94f269f-38955b7d;received=192.168.100.250
From: "327843803"sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200;tag=as2b6c28e1
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 1 INVITE
Server: FPBX-13.0.124(13.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:600852586@192.168.100.200:5060
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 534699741 534699741 IN IP4 192.168.100.200
s=Asterisk PBX 13.9.0
c=IN IP4 192.168.100.200
t=0 0
m=audio 17696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 044974, ts 000160, len 000160)
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 044975, ts 000320, len 000160)
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 044976, ts 000480, len 000160)
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 044977, ts 000640, len 000160)
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 044978, ts 000800, len 000160)
[2016-06-10 08:16:39] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 044979, ts 000960, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] chan_sip.c: Audio is at 17696
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] chan_sip.c: Adding codec ulaw to SDP
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.100.250:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-26216-94f269f-38955b7d;received=192.168.100.250
From: "327843803"sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200;tag=as2b6c28e1
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 1 INVITE
Server: FPBX-13.0.124(13.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:600852586@192.168.100.200:5060
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 534699741 534699741 IN IP4 192.168.100.200
s=Asterisk PBX 13.9.0
c=IN IP4 192.168.100.200
t=0 0
m=audio 17696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[2016-06-10 08:16:54] VERBOSE[2343] chan_sip.c:
<— SIP read from UDP:192.168.100.250:5060 —>
ACK sip:600852586@192.168.100.200:5060 SIP/2.0
From: sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200;tag=as2b6c28e1
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-26229-94f7326-5aec70c2
Max-Forwards: 70
Contact: sip:192.168.100.250:5060
Allow-Events: refer
Content-Length: 0
<------------->
[2016-06-10 08:16:54] VERBOSE[2343] chan_sip.c: — (10 headers 0 lines) —
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045732, ts 121440, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045733, ts 121600, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045734, ts 121760, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045735, ts 121920, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045736, ts 122080, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045737, ts 122240, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045738, ts 122400, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045739, ts 122560, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045740, ts 122720, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045741, ts 122880, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045742, ts 123040, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045743, ts 123200, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045744, ts 123360, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045745, ts 123520, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045746, ts 123680, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045747, ts 123840, len 000160)
[2016-06-10 08:16:54] VERBOSE[2343] chan_sip.c:
<— SIP read from UDP:192.168.100.250:5060 —>
INVITE sip:600852586@192.168.100.200:5060 SIP/2.0
From: sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200;tag=as2b6c28e1
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-2622a-94f7470-38658880
Max-Forwards: 70
Supported: replaces
Contact: sip:192.168.100.250:5060
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY
Allow-Events: refer
Content-Type: application/sdp
Content-Length: 233
v=0
o=Onnet 1300002000 100044578 IN IP4 192.168.100.250
s=1_CALLMEDIA
i=Phonetic
c=IN IP4 192.168.100.250
t=0 0
a=sendrecv
m=audio 49240 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[2016-06-10 08:16:54] VERBOSE[2343] chan_sip.c: — (13 headers 11 lines) —
[2016-06-10 08:16:54] VERBOSE[2343][C-00000004] chan_sip.c: Sending to 192.168.100.250:5060 (no NAT)
[2016-06-10 08:16:54] VERBOSE[2343][C-00000004] chan_sip.c:
<— Transmitting (no NAT) to 192.168.100.250:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-2622a-94f7470-38658880;received=192.168.100.250
From: sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200;tag=as2b6c28e1
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 2 INVITE
Server: FPBX-13.0.124(13.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:600852586@192.168.100.200:5060
Content-Length: 0
<------------>
[2016-06-10 08:16:54] VERBOSE[2343][C-00000004] chan_sip.c: Audio is at 17696
[2016-06-10 08:16:54] VERBOSE[2343][C-00000004] chan_sip.c: Adding codec ulaw to SDP
[2016-06-10 08:16:54] VERBOSE[2343][C-00000004] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2016-06-10 08:16:54] VERBOSE[2343][C-00000004] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.100.250:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-2622a-94f7470-38658880;received=192.168.100.250
From: sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200;tag=as2b6c28e1
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 2 INVITE
Server: FPBX-13.0.124(13.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:600852586@192.168.100.200:5060
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 534699741 534699741 IN IP4 192.168.100.200
s=Asterisk PBX 13.9.0
c=IN IP4 192.168.100.200
t=0 0
m=audio 17696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045748, ts 124000, len 000160)
[2016-06-10 08:16:54] VERBOSE[2343] chan_sip.c:
<— SIP read from UDP:192.168.100.250:5060 —>
ACK sip:600852586@192.168.100.200:5060 SIP/2.0
From: sip:192.168.100.250:5060;tag=20274190-fa64a8c0-13c4-50022-26216-2e3395f7-26216
To: sip:600852586@192.168.100.200;tag=as2b6c28e1
Call-ID: 1f853978-fa64a8c0-13c4-50022-26216-1bebc4c9-26216
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.100.250:5060;branch=z9hG4bK-2622a-94f747d-26b55dd2
Max-Forwards: 70
Contact: sip:192.168.100.250:5060
Allow-Events: refer
Content-Length: 0
<------------->
[2016-06-10 08:16:54] VERBOSE[2343] chan_sip.c: — (10 headers 0 lines) —
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000001, ts 369271308, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045749, ts 124160, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045750, ts 124320, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045751, ts 124480, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000002, ts 369271468, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000003, ts 369271628, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000004, ts 369271788, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045752, ts 124640, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000005, ts 369271948, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045753, ts 124800, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045754, ts 124960, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000006, ts 369272108, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000007, ts 369272268, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Sent RTP packet to 0.0.0.0:1234 (type 00, seq 045755, ts 125120, len 000160)
[2016-06-10 08:16:54] VERBOSE[3760][C-00000004] res_rtp_asterisk.c: Got RTP packet from 192.168.100.250:49240 (type 00, seq 000008, ts 369272428, len 000160)
etc.
How to correct it? I can’t change the CC server configuration.