403 forbidden outgoing call Proxy authetication problems

I have an Asterisk with internal extensions. On the other hand I have a SIP trunk contracted with my ISP which I use to take out incoming calls and make outgoing calls.

The problem is this.

I can receive calls: OK
But I can’t make calls: no

As I could check in a wireshark screenshot, when I make an external call from my extensions, Asterisk never indicates in the package “SIP INVITE”. My ISP requires that any outgoing calls be with SIP proxy authetication credentials.

Versions:
SNG7-PBX-64bit-2002
FreePBX 15.0.16.44
Asterisk Version: 16.9.0

This is my config:

Outcoming

type=peer
fromuser=+349XXXXXXXX
[email protected]
secret=
fromdomain=sip.isp.com
host=sip.isp.com
outboundproxy=proxy2.sip.isp.com
port=5060
nat=yes
insecure=port,invite
dtmfmode=auto
disallow=all
allow=ulaw,alaw
qualify=yes
keepalive=yes

Ahora vais a la pestaña Incoming.

En USER Context ponemos:

from trunk

En USER Details ponemos lo siguiente:

type=peer
username=+349XXXXXXXX
secret=
fromdomain=sip.isp.com
host=sip.isp.com
port=5060
outboundproxy=proxy2.sip.isp.com
qualify=no
nat=yes
insecure=port,invite
dtmfmode=auto
canreinvite=no
disallow=all
allow=ulaw,alaw
outboundproxyport=5060

Register String:

[email protected]::@proxy.sip.isp.com

STATUS Trunk ( registraction OK) (data changed for privacy)

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
ISP/+3490000000 5.5.5.5 Yes Yes 5060 OK (10 ms)
from trunk/+3490000000 5.5.5.5 Yes Yes 5060 Unmonitored
2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 0 offline]

PACKET CAPTURE (data changed for privacy)

SIP invite:

{INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.30.2:5160;branch=z9hG4bK00c6b487;rport
Max-Forwards: 70
From: sip:[email protected]:5160;tag=as28596747
To: sip:[email protected]:5060
Contact: sip:[email protected]:5160
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: CPD-1
Date: Sun, 29 Mar 2020 12:31:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 275
v=0
o=root 672991910 672991910 IN IP4 192.168.30.2
s=Asterisk PBX 16.9.0
c=IN IP4 192.168.30.2
t=0 0
m=audio 13248 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

Answer

SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.30.2:5160;received=192.168.30.2;rport=5160;branch=z9hG4bK00c6b487
To: sip:[email protected]:5060;tag=h7g4Esbg_qwqb4nczbggodqoixruhifv514iqm2s7
From: sip:[email protected]:5160;tag=as28596747
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

VERBOSE Log Asterisk

[2020-03-29 15:16:57] VERBOSE[20659][C-00000005] app_while.c: Jumping to priority 12
[2020-03-29 15:16:57] VERBOSE[20659][C-00000005] pbx.c: Executing [[email protected]:13] Return(“SIP/ISP-00000004”, “”) in new stack
[2020-03-29 15:16:57] VERBOSE[20659][C-00000005] app_stack.c: Spawn extension (from-trunk-sip-Isp, 60000000, 1) exited non-zero on ‘SIP/ISP-00000004’
[2020-03-29 15:16:57] VERBOSE[20659][C-00000005] app_stack.c: SIP/Isp-00000004 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2020-03-29 15:16:57] VERBOSE[20659][C-00000005] app_dial.c: Called SIP/isp/6000000000
[2020-03-29 15:16:57] WARNING[17098][C-00000005] chan_sip.c: Received response: “Forbidden” from ‘sip:[email protected]:5160;tag=as1be28a41’
[2020-03-29 15:16:57] VERBOSE[20659][C-00000005] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)

tests performed and negative result:

restart asterisk
reinstall the entire software
Change the value insecure=“port,invite” to “insecure=no”

Any idea how to force the SIP invite to always carry the SIP proxy authentication?

With insecure=no I haven’t been able to get the behaviour to change.

They aren’t challenging you for authentication via user/password. They are returning the 403 Forbidden immediately which means they aren’t using user/password to auth you and the IP your using isn’t whitelisted with them. If they are using user/password to auth you and this is the first response they give then they didn’t like how the initial request was sent to them. Either due to wrong details or missing details.

Hello,

Thank you for your interest,

The problem is that according to my ISP, I’m sending out call requests without a truncated header (sip proxy authentication) and then it rejects them.

The thing is that I don’t know how to configure Freepbx to always send SIP invites with header sip proxy authentication.

I don’t believe such a thing is possible. The Auth line on the invite must be based on the Auth line the provider sends back in the 401 Unauthorized packet.

1 Like

What he says it true. In order for you to auth with the other side, they must provide a 401 response with the realm and nonce to be used for authentication. So if they are expecting you to auth with a user/password scheme they’re failing to make that happen.

Okay, well, then what could be the problem? In the wireshark capture I see that the freepbx is registering correctly with the SIP server, but then when I want to make a call this happens.

You probably need to set Fromuser in the trunk to the same username you have for registration.

Hello,

I’ve tried what I was told and it’s still the same.

I have also captured the data with the ISP’s voice box that if the incoming and outgoing calls work. but I can’t replicate the behavior in asterisk.

ISP voice box:

äFÝc\Üæd¨E¸;@ùU5D£U>ôRÄÄ’ÕREGISTER sip:sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;rport;branch=z9hG4bK-4d9cc92e66748a775a32bcdc35248684
From: “+34900000000” sip:[email protected];tag=239579a726b299117f72
To: “+34900000000” sip:[email protected]
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, SUBSCRIBE, REFER, UPDATE
Max-Forwards: 70
Expires: 3600
User-Agent: Router CPE1
Content-Length: 0

\Üæd¨äFÝcE¸ y[,U>ôRU5D£ÄÄgfSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK-4d9cc92e66748a775a32bcdc35248684;rport=5060
From: “+34900000000” sip:[email protected];tag=239579a726b299117f72
To: “+34900000000” sip:[email protected];tag=1053d1a00cc839e8b0d312e9f76d8f26
Call-ID: [email protected]
CSeq: 1 REGISTER
Content-Length: 0
WWW-Authenticate: Digest nonce=“REQUESTHASH”,realm=“sip.isp.com”,algorithm=MD5,qop=“auth”
P-Charging-Function-Addresses: ccf=“aaa://sip.isp.com”
P-Charging-Vector: icid-value=1053d1a00cc839e8b0d312e9f7653766

äFÝc\Üæd¨E¸]@ÖU5D£U>ôRÄÄIÔREGISTER sip:sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;rport;branch=z9hG4bK-5150784a6045a5d16d1ec0d462c0db86
From: “+34900000000” sip:[email protected];tag=239579a726b299117f72
To: “+34900000000” sip:[email protected]
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: sip:[email protected]:5060
Authorization: Digest username="[email protected]", realm=“sip.isp.com”, nonce=“REQUESTHASH”, uri=“sip:sip.isp.com”, response=“RESPONDHASH”, algorithm=MD5, cnonce=“CNONCEHASH”, qop=auth, nc=00000001
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, SUBSCRIBE, REFER, UPDATE
Max-Forwards: 70
Expires: 3600
User-Agent: Router CPE1
Content-Length: 0

\Üæd¨äFÝcE¸yZFU>ôRU5D£ÄÄrÅ_SIP/2.0 200 OK
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK-5150784a6045a5d16d1ec0d462c0db86;rport=5060
From: “+34900000000” sip:[email protected];tag=239579a726b299117f72
To: “+34900000000” sip:[email protected];tag=1053d1a00cc851fc30d312e9f8036036
Call-ID: [email protected]
CSeq: 2 REGISTER
Content-Length: 0
Contact: sip:[email protected]:5060;expires=3600
P-Associated-URI: sip:[email protected]
P-Associated-URI: tel:+34900000000
P-Charging-Vector: icid-value=1053d1a00cc851fc30d312e9f7fe72cd;term-ioi=sip.isp.com
Authentication-Info: nextnonce=“HASH”,qop=auth,rspauth=“06592440ba254baf70f8ab15de59c08e”,cnonce=“CNONCEHASH”,nc=00000001
P-Charging-Function-Addresses: ccf=“aaa://sip.isp.com”

äFÝc\Üæd¨E¸@U5D£U>ôRÄÄ
&“INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK-31bd71df7168e01c2085ab75564ace33
From: “+34900000000” sip:[email protected];tag=26db5b4c2f8cda9139ddf6b5554c9182
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Proxy-Authorization: Digest username="[email protected]”, realm=“sip.isp.com”, nonce=“HASH”, uri="sip:[email protected]", response=“HASH1”, algorithm=MD5, cnonce=“HASH2”, qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, SUBSCRIBE, REFER, UPDATE
Max-Forwards: 70
User-Agent: Router CPE1
P-Asserted-Identity: “+34900000000” sip:[email protected]
Content-Length: 216

v=0
o=- 81747 2 IN IP4 22.22.22.22
s=-
c=IN IP4 22.22.22.22
t=0 0
m=audio 5050 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32
a=ptime:20

\Üæd¨äFÝcE¸My\U>ôRU5D£ÄÄ9¤æSIP/2.0 100 Trying
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK-31bd71df7168e01c2085ab75564ace33
From: “+34900000000” sip:[email protected];tag=26db5b4c2f8cda9139ddf6b5554c9182
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 INVITE

\Üæd¨äFÝcE¸ny[^U>ôRU5D£ÄÄZ SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK-31bd71df7168e01c2085ab75564ace33
From: “+34900000000” sip:[email protected];tag=26db5b4c2f8cda9139ddf6b5554c9182
To: sip:[email protected];tag=p65551t1537707424m343198c1149360512s1_1758194262-22192178
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0
Contact: sip:[email protected]:5060;transport=udp
Supported: timer
P-Charging-Vector: icid-value=016f19890c1398aa50d312ea15e87378
P-Charging-Function-Addresses: ccf=“aaa://sip.isp.com”

FREEPBX

äFÝcdÑTPþY#@E`v
?hZD7lU>ð(ÄbÿOPTIONS sip:sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 24.24.24.24:5160;branch=z9hG4bK3ea5ae19;rport
Max-Forwards: 70
From: “Unknown” sip:USER1234%[email protected]:5160;tag=as671b0ec6
To: sip:sip.isp.com
Contact: sip:USER1234%[email protected]:5160
Call-ID: [email protected]:5160
CSeq: 102 OPTIONS
User-Agent: Router CPE1
Date: Tue, 31 Mar 2020 11:08:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

)4ñmìTE Ó@5BU>ðÀ¨(
Ä(
þSIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.30.2:5160;received=192.168.30.2;rport=5160;branch=z9hG4bK3ea5ae19
To: sip:sip.isp.com;tag=h7g4Esbg_r3745vce929icv9b22li3tjgn25gp317
From: “Unknown” sip:USER1234%[email protected]:5160;tag=as671b0ec6
Call-ID: [email protected]:5160
CSeq: 102 OPTIONS
Content-Length: 0

mìT)4ñE`Ô
[email protected]?«À¨(
U>ð(ÄÀÊ REGISTER sip:sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 192.168.30.2:5160;branch=z9hG4bK015aa8cf;rport
Max-Forwards: 70
From: sip:[email protected];tag=as37f2eed8
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: Router CPE1
Expires: 3600
Contact: sip:[email protected]:5160
Content-Length: 0

dÑTPþYäFÝc£@E .
º@6äÈU>ðZD7lÄ(å1SIP/2.0 401 Unauthorized 11030230337
Via: SIP/2.0/UDP 24.24.24.24:5160;received=24.24.24.24;rport=5160;branch=z9hG4bK015aa8cf
To: sip:[email protected];tag=h7g4Esbg_05938f82082df05a224961e6bce
From: sip:[email protected];tag=as37f2eed8
Call-ID: [email protected]
CSeq: 102 REGISTER
Service-Route: sip:85.62.240.173:5060;transport=udp;lr
WWW-Authenticate: Digest realm=“sip.isp.com”,nonce=“96F93DDDA824835E0000000015ECF315”,algorithm=MD5,qop=“auth”
Content-Length: 0

mìT)4ñE`Ö
@>ˬ(
U>ð(ÄÂREGISTER sip:sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 192.168.30.2:5160;branch=z9hG4bK4a3c1ccd;rport
Max-Forwards: 70
From: sip:[email protected];tag=as37f2eed8
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 REGISTER
Supported: replaces, timer
User-Agent: Router CPE1
Authorization: Digest username="[email protected]", realm=“sip.isp.com”, algorithm=MD5, uri=“sip:sip.isp.com”, nonce=“96F93DDDA824835E0000000015ECF315”, response=“1a26f46219505f42c12081fc381dfd66”, qop=auth, cnonce=“49c9aff4”, nc=00000001
Expires: 3600
Contact: sip:[email protected]:5160
Content-Length: 0

dÑTPþYäFÝc£@E Ñ
ß@6äU>ðZD7lÄ(½ö0SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.24.24.24:5160;received=24.24.24.24;rport=5160;branch=z9hG4bK4a3c1ccd
To: sip:[email protected];tag=h7g4Esbg_05938f8204e1c05a224961f90b7
From: sip:[email protected];tag=as37f2eed8
Call-ID: [email protected]
CSeq: 103 REGISTER
Contact: sip:[email protected]:5160;expires=3600
P-Associated-Uri: sip:[email protected]
P-Associated-Uri: tel:+34900000000
Service-Route: sip:85.62.240.173:5060;transport=udp;lr
Content-Length: 0
Authentication-Info: nextnonce=“9EE2DBB7A824835E00000000ED6E2A29”,qop=auth,rspauth=“72955cd8f4a653bc65bd82dff2fd59ca”,cnonce=“49c9aff4”,nc=00000001

mìT)4ñE`¼AÑ@bÀ¨(
U>ð(ĨÍÂINVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.30.2:5160;branch=z9hG4bK6db2e2f0;rport
Max-Forwards: 70
From: sip:USER1234%[email protected]:5160;tag=as09317512
To: sip:[email protected]:5060
Contact: sip:USER1234%[email protected]:5160
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Router CPE1
Date: Tue, 31 Mar 2020 11:08:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1327473564 1327473564 IN IP4 192.168.30.2
s=Asterisk PBX 16.9.0
c=IN IP4 192.168.30.2
t=0 0
m=audio 10200 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

dÑTPþYäFÝc£@E ¤@@6ÆÚU>ðZD7lÄ(Ì"SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 24.24.24.24:5160;received=24.24.24.24;rport=5160;branch=z9hG4bK6db2e2f0
To: sip:[email protected]:5060;tag=h7g4Esbg_sfyvqn27hhjk48w1yb8nkeeksxcc2sqy
From: sip:USER1234%[email protected]:5160;tag=as09317512
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

Sorry, I was wrong about setting fromuser (at the time you hadn’t posted a capture from the ISP’s box). So you do want the phone number in From.

However, at the start of the call, the ISP box must be sending an INVITE without the Proxy-Authorization header. Normally, the sequence would be: box sends INVITE without Proxy-Authorization, Server sends 407, box sends ACK, box sends new INVITE with Proxy-Authorization.

I’d like to see a capture of the first INVITE from the box and compare it with the INVITE from FreePBX. With luck, we’ll see what is different that is causing the server to reject the latter.

I suspect an issue with Outbound Proxy and/or DNS resolution. For both the ISP box and FreePBX, please check the destination IP addresses for both REGISTER and INVITE and confirm that FreePBX is sending to the same addresses as the ISP’s box is using.

These are the first requests.

ISP VOX

äFÝc\Üæd¨E¸@U5D£U>ôRÄÄ
&“INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK-31bd71df7168e01c2085ab75564ace33
From: “+34900000000” sip:[email protected];tag=26db5b4c2f8cda9139ddf6b5554c9182
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Proxy-Authorization: Digest username="[email protected]”, realm=“sip.isp.com”, nonce=“HASH”, uri="sip:[email protected]", response=“HASH1”, algorithm=MD5, cnonce=“HASH2”, qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, SUBSCRIBE, REFER, UPDATE
Max-Forwards: 70
User-Agent: Router CPE1
P-Asserted-Identity: “+34900000000” sip:[email protected]
Content-Length: 216

FREEPBX

mìT)4ñE`¼AÑ@bÀ¨(
U>ð(ĨÍÂINVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.30.2:5160;branch=z9hG4bK6db2e2f0;rport
Max-Forwards: 70
From: sip:USER1234%40sip.isp.com@sip.isp.com:5160;tag=as09317512
To: sip:[email protected]:5060
Contact: sip:USER1234%40sip.isp.com@192.168.30.2:5160
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Router CPE1
Date: Tue, 31 Mar 2020 11:08:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277

I have checked it and the DNS resolution is the same in the ISP’s voice box as in FREEPBX.

Hello,

Maybe because freepbx uses port 5160 and the ISP voice box 5060, the problem is that I don’t know how to change freepbx’s port so that it doesn’t drive everything crazy.

Because I tried to change it from the gui and I had to restore the backup machine.

Disable your chan_sip trunk and create a pjsip trunk in FreePBX and see if it simply works?

Thank you very much for everything.

I just migrated the configuration to PJSIP to go native on port 5060.

I also tested and captured the traffic with another ISP voice box.

Right now the only differences I see are the following.

-freepbx always sends 5 register request, 3 with the original private ip and 2 with the public ip.

How can I send only one request with the public ip?

-there are more allow options in freepbx than in the voice box,

is there any way to disable Allow modes in freepbx?

-the branch and tag has different format in the isp voip and Freepbx.

How can I make them the same?

the sequence number starts at 21648 and in the isp box at 1 ,

how can you change so that the sequence always starts at 1 ?

freepbx is putting a parameter called line=clbrysr on the contact line but the ISP box is not.

How can I remove this parameter?

On freepbx there’s a line indicating Route but not in the voice box.

How can I remove the Route line::sip:sip.isp.com:5060 ?

FREEPBX

3x
mìT)4ñE`[email protected]@^À¨(
U>ôqÄÄ{ÛREGISTER sip:proxy2.sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 20.20.20.20:5060;rport;branch=z9hG4bKPj3aa87050-042d-4eea-a14c-b64239cf6f35
From: sip:[email protected];tag=a4054586-b6ce-4c5e-8b98-992cb265cd4e
To: sip:[email protected]
Call-ID: 60134c0f-8e0f-4125-aa66-96574f4eff07
CSeq: 21648 REGISTER
Contact: sip:[email protected]:5060;line=clbrysr
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Route: sip:sip.isp.com:5060
Max-Forwards: 70
User-Agent: ISP Voice box
Content-Length: 0

2x

äFÝcdÑTPþYE[email protected]ZD7lU>ôqÄÄ{2 REGISTER sip:proxy2.sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 20.20.20.20:5060;rport;branch=z9hG4bKPj3aa87050-042d-4eea-a14c-b64239cf6f35
From: sip:[email protected];tag=a4054586-b6ce-4c5e-8b98-992cb265cd4e
To: sip:[email protected]
Call-ID: 60134c0f-8e0f-4125-aa66-96574f4eff07
CSeq: 21648 REGISTER
Contact: sip:[email protected]:5060;line=clbrysr
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Route: sip:sip.isp.com:5060
Max-Forwards: 70
User-Agent: ISP Voice box
Content-Length: 0

dÑTPþYäFÝc£@E ¨ê@6ÞZU>ôqZD7lÄÄ×SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 20.20.20.20:5060;received=20.20.20.20;rport=5060;branch=z9hG4bKPj3aa87050-042d-4eea-a14c-b64239cf6f35
To: sip:[email protected];tag=wjsmxlcm2eunq6s2glohsxm1w
From: sip:[email protected];tag=a4054586-b6ce-4c5e-8b98-992cb265cd4e
Call-ID: 60134c0f-8e0f-4125-aa66-96574f4eff07
CSeq: 21648 REGISTER
Content-Length: 0

)4ñmìTE ªê@5VU>ôqÀ¨(
ÄÄSIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.30.2:5060;received=192.168.30.2;rport=5060;branch=z9hG4bKPj3aa87050-042d-4eea-a14c-b64239cf6f35
To: sip:[email protected];tag=wjsmxlcm2eunq6s2glohsxm1w
From: sip:[email protected];tag=a4054586-b6ce-4c5e-8b98-992cb265cd4e
Call-ID: 60134c0f-8e0f-4125-aa66-96574f4eff07
CSeq: 21648 REGISTER
Content-Length: 0

ISP

äFÝc\Üæd¨E¸;@ùU5D£U>ôRÄÄ’ÕREGISTER sip:sip.isp.com SIP/2.0
Via: SIP/2.0/UDP 20.20.20.20:5060;rport;branch=z9hG4bK-4d9cc92e66748a775a32bcdc35248684
From: “+34900000000” sip:[email protected];tag=239579a726b299117f72
To: “+34900000000” sip:[email protected]
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, SUBSCRIBE, REFER, UPDATE
Max-Forwards: 70
Expires: 3600
User-Agent: ISP Voice box
Content-Length: 0

\Üæd¨äFÝcE¸ y[,U>ôRU5D£ÄÄgfSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 20.20.20.20:5060;received=20.20.20.20;branch=z9hG4bK-4d9cc92e66748a775a32bcdc35248684;rport=5060
From: “+34900000000” sip:[email protected];tag=239579a726b299117f72
To: “+34900000000” sip:[email protected];tag=1053d1a00cc839e8b0d312e9f76d8f26
Call-ID: [email protected]
CSeq: 1 REGISTER
Content-Length: 0
WWW-Authenticate: Digest nonce=“HASH”,realm=“sip.isp.com”,algorithm=MD5,qop=“auth”
P-Charging-Function-Addresses: ccf=“aaa://sip.isp.com”
P-Charging-Vector: icid-value=1053d1a00cc839e8b0d312e9f7653766

can you help me?

normally the isp only verifies the name and username and password and user agent. but i don’t know how they identify any other value as well. Thank you.

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