Sip Connectivity Error


#1

Hello All,
Want to know some more in dept. When I am connecting softphone by using my mobile hotspot I am able to register myself successfully.
Below is the SIP DEBUG

<--- SIP read from UDP:223.191.57.215:9660 --->
REGISTER sip:88.99.245.202 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.100:5060;branch=z9hG4bK-524287-1---20959928a08a526b;rport
Max-Forwards: 70
Contact: <sip:Pokhraj@192.168.43.100:5060;rinstance=a12b41bbd3d2e635>;+sip.instance="<urn:uuid:7225f334-f594-53ff-aab2-e0303cf20e1d>";reg-id=1
To: "Pokhraj"<sip:Pokhraj@88.99.245.202>
From: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=c97f6f22
Call-ID: 94388MmZjMTYzZmM0NGJmNWJmMTMzNmI4ODU5ODRkNmRlYTI
CSeq: 1 REGISTER
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Supported: outbound, path
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 223.191.57.215:9660 (NAT)
Sending to 223.191.57.215:9660 (NAT)

<--- Transmitting (NAT) to 223.191.57.215:9660 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.43.100:5060;branch=z9hG4bK-524287-1---20959928a08a526b;received=223.191.57.215;rport=9660
From: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=c97f6f22
To: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=as73a14f18
Call-ID: 94388MmZjMTYzZmM0NGJmNWJmMTMzNmI4ODU5ODRkNmRlYTI
CSeq: 1 REGISTER
Server: Asterisk PBX 15.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5dbb8d86"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '94388MmZjMTYzZmM0NGJmNWJmMTMzNmI4ODU5ODRkNmRlYTI' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:223.191.57.215:9660 --->
REGISTER sip:88.99.245.202 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.100:5060;branch=z9hG4bK-524287-1---7c35b764bb684e1c;rport
Max-Forwards: 70
Contact: <sip:Pokhraj@192.168.43.100:5060;rinstance=a12b41bbd3d2e635>;+sip.instance="<urn:uuid:7225f334-f594-53ff-aab2-e0303cf20e1d>";reg-id=1
To: "Pokhraj"<sip:Pokhraj@88.99.245.202>
From: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=c97f6f22
Call-ID: 94388MmZjMTYzZmM0NGJmNWJmMTMzNmI4ODU5ODRkNmRlYTI
CSeq: 2 REGISTER
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Supported: outbound, path
User-Agent: X-Lite release 5.4.0 stamp 94388
Authorization: Digest username="Pokhraj",realm="asterisk",nonce="5dbb8d86",uri="sip:88.99.245.202",response="5e7de364ce823354a08ffd90519ff3ee",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 223.191.57.215:9660 (NAT)
    -- Registered SIP 'Pokhraj' at 223.191.57.215:9660
Reliably Transmitting (NAT) to 223.191.57.215:9660:
OPTIONS sip:Pokhraj@192.168.43.100:5060;rinstance=a12b41bbd3d2e635 SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK4290ce8f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as0c83c580
To: <sip:Pokhraj@192.168.43.100:5060;rinstance=a12b41bbd3d2e635>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 13aacd1c09b93a4a4b60d22625594745@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Mon, 11 Mar 2019 12:10:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 223.191.57.215:9660 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.43.100:5060;branch=z9hG4bK-524287-1---7c35b764bb684e1c;received=223.191.57.215;rport=9660
From: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=c97f6f22
To: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=as73a14f18
Call-ID: 94388MmZjMTYzZmM0NGJmNWJmMTMzNmI4ODU5ODRkNmRlYTI
CSeq: 2 REGISTER
Server: Asterisk PBX 15.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:Pokhraj@192.168.43.100:5060;rinstance=a12b41bbd3d2e635>;expires=3600
Date: Mon, 11 Mar 2019 12:10:01 GMT
Content-Length: 0

Now when I am trying to connect softphone by using the internet connection provided by my ISP from Home, I am getting error as below. Also I have purchased Public IP from my ISP which is 115.187.34.132

`Retransmitting #4 (NAT) to 223.191.57.215:9660:
OPTIONS sip:Pokhraj@223.191.57.215:9660;rinstance=a12b41bbd3d2e635 SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK7da47d52;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as775b70f6
To: <sip:Pokhraj@223.191.57.215:9660;rinstance=a12b41bbd3d2e635>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 7c7f61ad60ac3edf2ed55d9717e2b1c5@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Mon, 11 Mar 2019 12:22:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '7c7f61ad60ac3edf2ed55d9717e2b1c5@88.99.245.202:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 223.191.57.215:9660:
OPTIONS sip:Pokhraj@223.191.57.215:9660;rinstance=a12b41bbd3d2e635 SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK6f876ee7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as32a7ac20
To: <sip:Pokhraj@223.191.57.215:9660;rinstance=a12b41bbd3d2e635>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 47d350a45954e96f103603b25a7f290a@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Mon, 11 Mar 2019 12:23:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 223.191.57.215:9660:
OPTIONS sip:Pokhraj@223.191.57.215:9660;rinstance=a12b41bbd3d2e635 SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK6f876ee7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as32a7ac20
To: <sip:Pokhraj@223.191.57.215:9660;rinstance=a12b41bbd3d2e635>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 47d350a45954e96f103603b25a7f290a@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Mon, 11 Mar 2019 12:23:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0`

Could anyone help me to figure out the issue please


#2

You neither de-registered from the mobile hot-spot, nor re-registered via the ISP, so Asterisk assumes you are still best reached via the mobile hot spot, but, as you are no longer there, is connectivity checks fail.

If there is no evidence of registrations via the ISP, your ISP may be blocking SIP, or your router may be doing so. You will need to find where, in the network, the REGISTER request gets lost.

Note that SIP phone may also not have realised that its IP address has changed.


#3

Absolutely correct. I have tracert the details and put all the details to my ISP.
They have changed the IP and unblock the port 5060 and now I am able to connect successfully with all the peers.
Thanks once again for your valuable advice.