I setup a DID number to receive calls and it works, but sometimes the number stays busy.
When the calls work the server registration shows the information about the number and logs, when it doesn’t work the server doesn’t register anything.
Please someone help me. How can I monitor it.
Could be problem of my supplier of DID number?
Thank you.
Leandro
Registration is used when you dont have static IP, or you cant route the DI calls to a SIP uri, you should enable SIP debug when you re not able to register your SIP account with your DID provider, also ask them if there is some issue at the moment it fails
Thank you ambiorixg12, I will leave the sip debug enable and enter in contact with my provider.
Hello everyone,
I tested and the problem is happening when I try to receive a call with the internet of my carrier in Brazil - Telefonica VIVO… Anyone suspect what is this? Carrier normally block ports to receive calls in the port 5060?
Bellow can see the debug of a test call:
== Using SIP RTP CoS mark 5
– Executing [551935177554@incoming:1] Dial(“SIP/buydid-1-00000014”, “SIP/5000,180”) in new stack
[Jan 26 09:12:39] WARNING[23270][C-0000006c]: app_dial.c:2525 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/buydid-1-00000014’ status is ‘CHANUNAVAIL’
Reliably Transmitting (NAT) to 177.79.71.254:54997:
OPTIONS sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK398a0f73;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@54.94.200.147;tag=as2d77824a
To: sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP
Contact: sip:asterisk@54.94.200.147:5060
Call-ID: 0266758f34998e5402ed94265f5146b2@54.94.200.147:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 26 Jan 2018 11:12:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
== Using SIP RTP CoS mark 5
– Executing [551935177554@incoming:1] Dial(“SIP/buydid-2-00000015”, “SIP/5000,180”) in new stack
[Jan 26 09:12:39] WARNING[23271][C-0000006d]: app_dial.c:2525 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/buydid-2-00000015’ status is ‘CHANUNAVAIL’<— SIP read from UDP:177.79.71.254:54997 —>
REGISTER sip:solaristelecom.ddns.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 177.79.71.254:54997;branch=z9hG4bK-524287-1—251e61ea74aa0be1;rport
Max-Forwards: 70
Contact: sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP
To: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP
From: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP;tag=bf43af7e
Call-ID: ZoL5faaZzfwoR2Q-TIowdw…
CSeq: 43 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rd82a609
Authorization: Digest username=“5000”,realm=“asterisk”,nonce=“5d66aa4b”,uri=“sip:solaristelecom.ddns.net;transport=UDP”,response=“4684d9ee1c65ce00670551a7929dd583”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0<------------->
— (14 headers 0 lines) —
Sending to 177.79.71.254:54997 (NAT)
Sending to 177.79.71.254:54997 (NAT)<— Transmitting (NAT) to 177.79.71.254:54997 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 177.79.71.254:54997;branch=z9hG4bK-524287-1—251e61ea74aa0be1;received=177.79.71.254;rport=54997
From: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP;tag=bf43af7e
To: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP;tag=as61dc7854
Call-ID: ZoL5faaZzfwoR2Q-TIowdw…
CSeq: 43 REGISTER
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“00d40e58”
Content-Length: 0<------------>
Scheduling destruction of SIP dialog ‘ZoL5faaZzfwoR2Q-TIowdw…’ in 32000 ms (Method: REGISTER)<— SIP read from UDP:177.79.71.254:54997 —>
REGISTER sip:solaristelecom.ddns.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 177.79.71.254:54997;branch=z9hG4bK-524287-1—586cd9a26f140996;rport
Max-Forwards: 70
Contact: sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP
To: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP
From: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP;tag=bf43af7e
Call-ID: ZoL5faaZzfwoR2Q-TIowdw…
CSeq: 44 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rd82a609
Authorization: Digest username=“5000”,realm=“asterisk”,nonce=“00d40e58”,uri=“sip:solaristelecom.ddns.net;transport=UDP”,response=“912fb9e05bcee04d13cce57a5ae4cc38”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0<------------->
— (14 headers 0 lines) —
Sending to 177.79.71.254:54997 (NAT)
[Jan 26 09:12:39] NOTICE[16835]: chan_sip.c:30192 sip_poke_peer: Still have a QUALIFY dialog active, deleting
Reliably Transmitting (NAT) to 177.79.71.254:54997:
OPTIONS sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK0db86d71;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@54.94.200.147;tag=as4bb5af0a
To: sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP
Contact: sip:asterisk@54.94.200.147:5060
Call-ID: 488ec3142357c365191e10fa48b1374b@54.94.200.147:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 26 Jan 2018 11:12:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— Transmitting (NAT) to 177.79.71.254:54997 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 177.79.71.254:54997;branch=z9hG4bK-524287-1—586cd9a26f140996;received=177.79.71.254;rport=54997
From: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP;tag=bf43af7e
To: "Leandro CEL"sip:5000@solaristelecom.ddns.net;transport=UDP;tag=as61dc7854
Call-ID: ZoL5faaZzfwoR2Q-TIowdw…
CSeq: 44 REGISTER
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP;expires=60
Date: Fri, 26 Jan 2018 11:12:39 GMT
Content-Length: 0<------------>
Scheduling destruction of SIP dialog ‘ZoL5faaZzfwoR2Q-TIowdw…’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘0266758f34998e5402ed94265f5146b2@54.94.200.147:5060’ Method: OPTIONS
Retransmitting #1 (NAT) to 177.79.71.254:54997:
OPTIONS sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK0db86d71;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@54.94.200.147;tag=as4bb5af0a
To: sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP
Contact: sip:asterisk@54.94.200.147:5060
Call-ID: 488ec3142357c365191e10fa48b1374b@54.94.200.147:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 26 Jan 2018 11:12:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Retransmitting #2 (NAT) to 177.79.71.254:54997:
OPTIONS sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK0db86d71;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@54.94.200.147;tag=as4bb5af0a
To: sip:5000@177.79.71.254:54997;rinstance=40f3933d9c5dfbdc;transport=UDP
Contact: sip:asterisk@54.94.200.147:5060
Call-ID: 488ec3142357c365191e10fa48b1374b@54.94.200.147:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 26 Jan 2018 11:12:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
There is call in your log.
Yes david, but the call didn’t ring in my smartphone when I use the internet provider from my 4g network.
For some reason Asterisk cant reach your device 5000 , which I assumed it is registered on your Smart phone, make sure your 4G provider is not causing the issue
ambiorxg12,
I evolved a little bit in this issue, I discover that my number is becoming unreacheble when I am using my 4G network.
If the parameter qualify=yes stay unreachable. If I put to qualify=no the reception works normally.
Have an idea to resolve it?
It is some kind of ping method, where Asterisk send option request to a remote peer to verify it is rechable,
Put qualify=no!!!
Or fix the peer so it handles OPTIONs according to the RFC (i.e. it provides a response of some kind, even an error one).
I understood, but with your experience. Do you think that I should to try to resolve it with my provider, or could be worst?
I thinked that was the problem solved when I put qualify=no for all clients, but I saw that I was wrong. After I did it the incoming calls of my did number is not receiving, register in the log server, but doesn’t call the telephone IP. Anyone can explain why it works this.
In this case to me qualify can’t be “no” because can leave the destionations doesn’t call, when the number is available, and ok.
That leaves fix the peer.
Well, in the truth the problem persist, I tried with qualify “no and yes” and no one works stabling.
Sometimes the calls arrived to my server and doesn’t ring the extensions, sometimes doesn’t looks nothing in the server and for last sometimes the calls ring and works perfectly.
Can you check my dialplan please, maybe have some mistakes that can to resolve it, thank you to all:
>
> [basico]
> exten => _XXXXXXX,1,Dial(SIP/${EXTEN},60})
> exten => _XXXXXXX,2,Answer()
> exten => _XXXXXXX,3,Hangup()
> include = messages
>
> [incoming]
> exten => 551935177554,1,Dial(SIP/4701006,60) ; direct inboound call from DID provider to specific extention
> same => n,ExecIf([${HANGUPCAUSE}=19]?Dial(SIP/voxbeam-solaris/00111025519999901569,120)) ; direct inboound call from DID provider to specific extention
> same => n,Playtones(congestion)
> same => n,Congestion(5)
> same => n,Hangup()
>
> [rotadesaida-silver] ;Silver
> exten => _+XXXXXXXX.,1,Dial(SIP/voxbeam-solaris/0011103${EXTEN:1},180/nj)
> same => n,ExecIf([${HANGUPCAUSE}=34]?Dial(SIP/voxbeam-solaris/0011102${EXTEN:1},180/nj))
> exten => _XXXXXXXX.,1,Dial(SIP/voxbeam-solaris/0011103${EXTEN},180/nj)
> same => n,ExecIf([${HANGUPCAUSE}=34]?Dial(SIP/voxbeam-solaris/0011102${EXTEN},180/nj))
> include = basico
> include = incoming
> include = messages
>
> [rotadesaida-platinum]
> exten => _+XXXXXXXX.,1,Dial(SIP/voxbeam-solaris/0011102${EXTEN:1},180/nj)
> exten => _XXXXXXXX.,1,Dial(SIP/voxbeam-solaris/0011102${EXTEN},180/nj)
> include = basico
> include = incoming
> include = messages
>
> [rotadesaida-gold]
> exten => _+XXXXXXXX.,1,Dial(SIP/voxbeam-solaris/0011101${EXTEN:1},180/nj)
> exten => _XXXXXXXX.,1,Dial(SIP/voxbeam-solaris/0011101${EXTEN},180/nj)
> include = basico
> include = incoming
> include = messages
>
> [messages]
> exten => _XXXXXXX,1,MessageSend(sip:${EXTEN},"${CALLERID(name)}"${MESSAGE(from)})
>
> [BLF_Solaris]
> exten => _XXXXXXX,hint,SIP/${EXTEN}
Turn up the logging to see why the qualifies are failing. Maybe the round trip time you are requiring is not achievable in your system.
Follow the answer:
Did work perfectly, very fast answer to ring the destination (2 to 5 seconds):
== Using SIP RTP CoS mark 5
-- Executing [551935177554@incoming:1] Dial("SIP/buydid-1-00000011", "SIP/4701006,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/4701006
-- SIP/4701006-00000012 is ringing
[Feb 2 09:53:20] NOTICE[20873]: chan_sip.c:28633 handle_request_register: Registration from <sip:8007@solaristelecom.ddns.net;transport=UDP>' failed for '187.107.128.238:29358' - Wrong password
== Spawn extension (incoming, 551935177554, 1) exited non-zero on 'SIP/buydid-1-00000011'
Flow:
Didn’t work:
== Using SIP RTP CoS mark 5
-- Executing [551935177554@incoming:1] Dial("SIP/buydid-1-0000003c", "SIP/4701006,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/4701006
== Spawn extension (incoming, 551935177554, 1) exited non-zero on 'SIP/buydid-1-0000003c'
[Feb 2 13:46:35] NOTICE[20873]: chan_sip.c:28633 handle_request_register: Registration from '<sip:8009@solaristelecom.ddns.net;transport=UDP>' failed for '187.107.128.238:28216' - Wrong password
-- Registered SIP '4701006' at 181.221.135.180:4100
Flow:
… after the call fall.
Adding the log of DEBUG mode:
== Using SIP RTP CoS mark 5
-- Executing [551935177554@incoming:1] Dial("SIP/buydid-1-00000040", "SIP/4701006,60") in new stack
== Using SIP RTP CoS mark 5
Audio is at 17934
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 181.221.135.180:6916:
INVITE sip:4701006@192.168.137.220:5060 SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK4969179b;rport
Max-Forwards: 70
From: "5519999901569" <sip:5519999901569@54.94.200.147>;tag=as1dfa9f57
To: <sip:4701006@192.168.137.220:5060>
Contact: <sip:5519999901569@54.94.200.147:5060>
Call-ID: 4699d8c57ec41639218d474f2c3e2c5a@54.94.200.147:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 02 Feb 2018 17:28:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:+551935177554@54.94.200.147>;reason=unconditional
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1758798933 1758798933 IN IP4 54.94.200.147
s=Asterisk PBX 13.18.5
c=IN IP4 54.94.200.147
t=0 0
m=audio 17934 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
-- Called SIP/4701006
Retransmitting #1 (NAT) to 181.221.135.180:6916:
INVITE sip:4701006@192.168.137.220:5060 SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK4969179b;rport
Max-Forwards: 70
From: "5519999901569" <sip:5519999901569@54.94.200.147>;tag=as1dfa9f57
To: <sip:4701006@192.168.137.220:5060>
Contact: <sip:5519999901569@54.94.200.147:5060>
Call-ID: 4699d8c57ec41639218d474f2c3e2c5a@54.94.200.147:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 02 Feb 2018 17:28:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:+551935177554@54.94.200.147>;reason=unconditional
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1758798933 1758798933 IN IP4 54.94.200.147
s=Asterisk PBX 13.18.5
c=IN IP4 54.94.200.147
t=0 0
m=audio 17934 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
Retransmitting #2 (NAT) to 181.221.135.180:6916:
INVITE sip:4701006@192.168.137.220:5060 SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK4969179b;rport
Max-Forwards: 70
From: "5519999901569" <sip:5519999901569@54.94.200.147>;tag=as1dfa9f57
To: <sip:4701006@192.168.137.220:5060>
Contact: <sip:5519999901569@54.94.200.147:5060>
Call-ID: 4699d8c57ec41639218d474f2c3e2c5a@54.94.200.147:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 02 Feb 2018 17:28:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:+551935177554@54.94.200.147>;reason=unconditional
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1758798933 1758798933 IN IP4 54.94.200.147
s=Asterisk PBX 13.18.5
c=IN IP4 54.94.200.147
t=0 0
m=audio 17934 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
Retransmitting #3 (NAT) to 181.221.135.180:6916:
INVITE sip:4701006@192.168.137.220:5060 SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK4969179b;rport
Max-Forwards: 70
From: "5519999901569" <sip:5519999901569@54.94.200.147>;tag=as1dfa9f57
To: <sip:4701006@192.168.137.220:5060>
Contact: <sip:5519999901569@54.94.200.147:5060>
Call-ID: 4699d8c57ec41639218d474f2c3e2c5a@54.94.200.147:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 02 Feb 2018 17:28:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:+551935177554@54.94.200.147>;reason=unconditional
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1758798933 1758798933 IN IP4 54.94.200.147
s=Asterisk PBX 13.18.5
c=IN IP4 54.94.200.147
t=0 0
m=audio 17934 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
Retransmitting #4 (NAT) to 181.221.135.180:6916:
INVITE sip:4701006@192.168.137.220:5060 SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK4969179b;rport
Max-Forwards: 70
From: "5519999901569" <sip:5519999901569@54.94.200.147>;tag=as1dfa9f57
To: <sip:4701006@192.168.137.220:5060>
Contact: <sip:5519999901569@54.94.200.147:5060>
Call-ID: 4699d8c57ec41639218d474f2c3e2c5a@54.94.200.147:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 02 Feb 2018 17:28:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:+551935177554@54.94.200.147>;reason=unconditional
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1758798933 1758798933 IN IP4 54.94.200.147
s=Asterisk PBX 13.18.5
c=IN IP4 54.94.200.147
t=0 0
m=audio 17934 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
[Feb 2 15:28:13] NOTICE[20873]: chan_sip.c:28633 handle_request_register: Registration from '<sip:8009@solaristelecom.ddns.net;transport=UDP>' failed for '138.36.212.181:10535' - Wrong password
Retransmitting #5 (NAT) to 181.221.135.180:6916:
INVITE sip:4701006@192.168.137.220:5060 SIP/2.0
Via: SIP/2.0/UDP 54.94.200.147:5060;branch=z9hG4bK4969179b;rport
Max-Forwards: 70
From: "5519999901569" <sip:5519999901569@54.94.200.147>;tag=as1dfa9f57
To: <sip:4701006@192.168.137.220:5060>
Contact: <sip:5519999901569@54.94.200.147:5060>
Call-ID: 4699d8c57ec41639218d474f2c3e2c5a@54.94.200.147:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.5
Date: Fri, 02 Feb 2018 17:28:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:+551935177554@54.94.200.147>;reason=unconditional
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1758798933 1758798933 IN IP4 54.94.200.147
s=Asterisk PBX 13.18.5
c=IN IP4 54.94.200.147
t=0 0
m=audio 17934 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
Scheduling destruction of SIP dialog '4699d8c57ec41639218d474f2c3e2c5a@54.94.200.147:5060' in 32000 ms (Method: INVITE)
== Spawn extension (incoming, 551935177554, 1) exited non-zero on 'SIP/buydid-1-00000040'
Problem identified, it’s necessary to use qualify=yes to receive calls behind a NAT, and in my mobile phone the client is unreachable, I have a galaxy S6 and I don’t know why, the same carrier works in an iphone 6S.
The necessity arises from the NAT implementation. It is likely not that you have to send OPTIONS, but that you have to send some traffic. Using a short register refresh interval might work as well.
Interesting david, have others possibilities to do it different of the qualify uses?
Thank you!