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I have a home server in my local network with Asterisk installed. It has been working without any problem for several month. I have just changed my router (ZyXel Keenetic Lite -> LP-LINK TL-R480T+) and Asterisk stop working correct. If there is no outbound calls for at least 5 minutes Asterisk starts recieving 403 from provider on every new outbound call. So every time I want to use my phone I must type "sip reload" in CLI.

Interesting facts:

  1. Branded provider's program works well (so it is not a provider's problem)
  2. Asterisk still works good with old router
  3. The request that provider rejects is the same that provider accepts: means only delay from the last call

What I have already tried:

  1. Add server to DMZ
  2. Enable\Disable SIP ALG
  3. Change some asterisk settings

Here is debug logs:

403 case:

<--- SIP read from UDP:192.168.1.33:5060 --->
INVITE sip:100@192.168.1.10 SIP/2.0
Call-ID: b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=8648d1de
To: <sip:100@192.168.1.10>
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-3938-3062199cc37c8b1eb032f59ed8056574
Max-Forwards: 70
Contact: "Vasiliy" <sip:polik94@192.168.1.33:5060;transport=udp;registering_acc=192_168_1_10>
User-Agent: Jitsi2.8.5426Windows 7
Content-Type: application/sdp
Content-Length: 940

v=0
o=polik94-jitsi.org 0 0 IN IP4 192.168.1.33
s=-
c=IN IP4 192.168.1.33
t=0 0
m=audio 5097 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5099 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[0-1920],y=[0-1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[0-1920],y=[0-1080]]
<------------->
--- (11 headers 32 lines) ---
Sending to 192.168.1.33:5060 (NAT)
Using INVITE request as basis request - b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
Found peer 'polik94' for 'polik94' from 192.168.1.33:5060

<--- Reliably Transmitting (NAT) to 192.168.1.33:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-3938-3062199cc37c8b1eb032f59ed8056574;received=192.168.1.33;rport=5060
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=8648d1de
To: <sip:100@192.168.1.10>;tag=as5b98de20
Call-ID: b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c52ef35"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.33:5060 --->
ACK sip:100@192.168.1.10 SIP/2.0
Call-ID: b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=8648d1de
To: <sip:100@192.168.1.10>;tag=as5b98de20
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-3938-3062199cc37c8b1eb032f59ed8056574
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.33:5060 --->
INVITE sip:100@192.168.1.10 SIP/2.0
Call-ID: b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=8648d1de
To: <sip:100@192.168.1.10>
Max-Forwards: 70
Contact: "Vasiliy" <sip:polik94@192.168.1.33:5060;transport=udp;registering_acc=192_168_1_10>
User-Agent: Jitsi2.8.5426Windows 7
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-3938-da5c355d81b7fa5083d9c44fe7f7db06
Authorization: Digest username="polik94",realm="asterisk",nonce="3c52ef35",uri="sip:100@192.168.1.10",response="e2fd92b7d377ca126b8caf7718ea1ee7",algorithm=MD5
Content-Length: 940

v=0
o=polik94-jitsi.org 0 0 IN IP4 192.168.1.33
s=-
c=IN IP4 192.168.1.33
t=0 0
m=audio 5097 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5099 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[0-1920],y=[0-1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[0-1920],y=[0-1080]]
<------------->
--- (12 headers 32 lines) ---
Sending to 192.168.1.33:5060 (NAT)
Using INVITE request as basis request - b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
Found peer 'polik94' for 'polik94' from 192.168.1.33:5060
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 9
Found RTP audio format 100
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 103
Found RTP audio format 3
Found RTP audio format 104
Found RTP audio format 4
Found RTP audio format 101
Found unknown media description format opus for ID 96
Found unknown media description format SILK for ID 97
Found unknown media description format SILK for ID 98
Found audio description format G722 for ID 9
Found unknown media description format speex for ID 100
Found audio description format speex for ID 102
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 103
Found audio description format GSM for ID 3
Found audio description format speex for ID 104
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Found RTP video format 105
Found RTP video format 99
Found video description format H264 for ID 105
Found video description format H264 for ID 99
Capabilities: us - 0x8 (alaw), peer - audio=0x20000160f (g723|gsm|ulaw|alaw|speex|speex16|ilbc|g722)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.33:5097
Looking for 100 in polik94 (domain 192.168.1.10)
list_route: hop: <sip:polik94@192.168.1.33:5060;transport=udp;registering_acc=192_168_1_10>

<--- Transmitting (NAT) to 192.168.1.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-3938-da5c355d81b7fa5083d9c44fe7f7db06;received=192.168.1.33;rport=5060
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=8648d1de
To: <sip:100@192.168.1.10>
Call-ID: b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:100@192.168.1.10:5060>
Content-Length: 0


<------------>
Audio is at 13450
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to [provider's ip]:5060:
INVITE sip:100@sip.provider SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5d12069d;rport
Max-Forwards: 70
From: "Polik94" <sip:qetuosf13@sip.provider>;tag=as1e6dfa83
To: <sip:100@sip.provider>
Contact: <sip:qetuosf13@192.168.1.10:5060>
Call-ID: 17b73fdd6883e4aa5de8279158a03d15@sip.provider
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Date: Mon, 07 Sep 2015 10:05:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1488403740 1488403740 IN IP4 192.168.1.10
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 192.168.1.10
t=0 0
m=audio 13450 RTP/AVP 8 101
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:10.25.0.50:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.60.97.44;rport=3681;branch=z9hG4bK5d12069d
To: <sip:100@sip.provider>;tag=h7g4Esbg_24ixe2ek6vyekgw2o5r1j7ovzgv3ucr3
From: "Polik94" <sip:qetuosf13@sip.provider>;tag=as1e6dfa83
Call-ID: 17b73fdd6883e4aa5de8279158a03d15@sip.provider
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
set_destination: Parsing <sip:100@sip.provider> for address/port to send to
set_destination: set destination to [provider's ip]:5060
Transmitting (NAT) to 10.25.0.50:5060:
ACK sip:100@sip.provider SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5d12069d;rport
Max-Forwards: 70
From: "Polik94" <sip:qetuosf13@sip.provider>;tag=as1e6dfa83
To: <sip:100@sip.provider>;tag=h7g4Esbg_24ixe2ek6vyekgw2o5r1j7ovzgv3ucr3
Contact: <sip:qetuosf13@192.168.1.10:5060>
Call-ID: 17b73fdd6883e4aa5de8279158a03d15@sip.provider
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Content-Length: 0


---
[Sep  7 13:05:47] WARNING[7768]: chan_sip.c:20366 handle_response_invite: Received response: "Forbidden" from '"Polik94" <sip:qetuosf13@sip.provider>;tag=as1e6dfa83'
Scheduling destruction of SIP dialog '17b73fdd6883e4aa5de8279158a03d15@sip.provider' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog 'b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.1.33:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-3938-da5c355d81b7fa5083d9c44fe7f7db06;received=192.168.1.33;rport=5060
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=8648d1de
To: <sip:100@192.168.1.10>;tag=as456d5c1c
Call-ID: b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.33:5060 --->
ACK sip:100@192.168.1.10 SIP/2.0
Call-ID: b4dd546d9cc5e887e18edfe27e94cdcc@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=8648d1de
To: <sip:100@192.168.1.10>;tag=as456d5c1c
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-3938-da5c355d81b7fa5083d9c44fe7f7db06
CSeq: 2 ACK
Content-Length: 0

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

normal case:

 <--- SIP read from UDP:192.168.1.33:5060 --->
INVITE sip:100@192.168.1.10 SIP/2.0
Call-ID: 2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=ab31307c
To: <sip:100@192.168.1.10>
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-333535-28ca8acdf12ffd3de8c300ccb14e6191
Max-Forwards: 70
Contact: "Vasiliy" <sip:polik94@192.168.1.33:5060;transport=udp;registering_acc=192_168_1_10>
User-Agent: Jitsi2.8.5426Windows 7
Content-Type: application/sdp
Content-Length: 940

v=0
o=polik94-jitsi.org 0 0 IN IP4 192.168.1.33
s=-
c=IN IP4 192.168.1.33
t=0 0
m=audio 5008 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5010 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[0-1920],y=[0-1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[0-1920],y=[0-1080]]
<------------->
--- (11 headers 32 lines) ---
Sending to 192.168.1.33:5060 (NAT)
Using INVITE request as basis request - 2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0
Found peer 'polik94' for 'polik94' from 192.168.1.33:5060

<--- Reliably Transmitting (NAT) to 192.168.1.33:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-333535-28ca8acdf12ffd3de8c300ccb14e6191;received=192.168.1.33;rport=5060
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=ab31307c
To: <sip:100@192.168.1.10>;tag=as56a44415
Call-ID: 2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27282bd1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.33:5060 --->
ACK sip:100@192.168.1.10 SIP/2.0
Call-ID: 2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=ab31307c
To: <sip:100@192.168.1.10>;tag=as56a44415
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-333535-28ca8acdf12ffd3de8c300ccb14e6191
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.33:5060 --->
INVITE sip:100@192.168.1.10 SIP/2.0
Call-ID: 2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=ab31307c
To: <sip:100@192.168.1.10>
Max-Forwards: 70
Contact: "Vasiliy" <sip:polik94@192.168.1.33:5060;transport=udp;registering_acc=192_168_1_10>
User-Agent: Jitsi2.8.5426Windows 7
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-333535-d1bf83f1b060e7c4b514bd35cb6dff80
Authorization: Digest username="polik94",realm="asterisk",nonce="27282bd1",uri="sip:100@192.168.1.10",response="6de03926844af3f3ca92df2567f4e59c",algorithm=MD5
Content-Length: 940

v=0
o=polik94-jitsi.org 0 0 IN IP4 192.168.1.33
s=-
c=IN IP4 192.168.1.33
t=0 0
m=audio 5008 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5010 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[0-1920],y=[0-1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[0-1920],y=[0-1080]]
<------------->
--- (12 headers 32 lines) ---
Sending to 192.168.1.33:5060 (NAT)
Using INVITE request as basis request - 2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0
Found peer 'polik94' for 'polik94' from 192.168.1.33:5060
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 9
Found RTP audio format 100
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
ound RTP audio format 103
Found RTP audio format 3
Found RTP audio format 104
Found RTP audio format 4
Found RTP audio format 101
Found unknown media description format opus for ID 96
Found unknown media description format SILK for ID 97
Found unknown media description format SILK for ID 98
Found audio description format G722 for ID 9
Found unknown media description format speex for ID 100
Found audio description format speex for ID 102
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 103
Found audio description format GSM for ID 3
Found audio description format speex for ID 104
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Found RTP video format 105
Found RTP video format 99
Found video description format H264 for ID 105
Found video description format H264 for ID 99
Capabilities: us - 0x8 (alaw), peer - audio=0x20000160f (g723|gsm|ulaw|alaw|speex|speex16|ilbc|g722)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.33:5008
Looking for 100 in polik94 (domain 192.168.1.10)
list_route: hop: <sip:polik94@192.168.1.33:5060;transport=udp;registering_acc=192_168_1_10>

<--- Transmitting (NAT) to 192.168.1.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-333535-d1bf83f1b060e7c4b514bd35cb6dff80;received=192.168.1.33;rport=5060
From: "Vasiliy" <sip:polik94@192.168.1.10>;tag=ab31307c
To: <sip:100@192.168.1.10>
Call-ID: 2f3d3686ace5057c7daa94e1320a64b0@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:100@192.168.1.10:5060>
Content-Length: 0


<------------>
Audio is at 10708
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to [provider's ip]:5060:
INVITE sip:100@sip.provider SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK165014c3;rport
Max-Forwards: 70
From: "Polik94" <sip:qetuosf13@sip.provider>;tag=as155f43ea
To: <sip:100@sip.provider>
Contact: <sip:qetuosf13@192.168.1.10:5060>
Call-ID: 4184da463f6b547f6ff476fe178f1102@sip.provider
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Date: Sat, 05 Sep 2015 18:42:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1206881313 1206881313 IN IP4 192.168.1.10
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 192.168.1.10
t=0 0
m=audio 10708 RTP/AVP 8 101
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:10.25.0.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.60.97.44;rport=56577;branch=z9hG4bK165014c3
To: <sip:100@sip.provider>
From: "Polik94" <sip:qetuosf13@sip.provider>;tag=as155f43ea
Call-ID: 4184da463f6b547f6ff476fe178f1102@sip.provider
CSeq: 102 INVITE
Content-Length: 0

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

<...>
share|improve this question
    
Is it a trunk with registration? I think when you are trying to make a call, it turns out that a trunk is not registered. What shows sip show registry in asterisk cli before a 403 case call? – Sergey S. Sep 7 '15 at 15:29
    
The state is "Registered" just before a call. And the same just after. – preveter Sep 7 '15 at 17:17

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