Authorizing to service provider

Asterisk Asterisk SIP
1

Hi there,

I am quite new to Asterisk, setting up my first system. I am trying to make a PBX on our network with asterisk on an Ubuntu18 vm.
Internal calling and inbound calls go just fine.
However, when attempting to make an outbound call I first get a 100 trying and then a 403 forbidden.

My configuration is basically the sample config from https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples (no.2)

<--- Transmitting SIP request (901 bytes) to UDP:[myprovider]:5060 --->
INVITE sip:[mobilephone]@sip.myprovider.com:5060 SIP/2.0
Via: SIP/2.0/UDP [myip]:5060;rport;branch=z9hG4bKPjTCmUWug.r6T.8eq.0d4NuM4QyegnESSy
From: "[deskphone]" <sip:[localnumber]@192.168.1.3>;tag=jHA.w0JCu.Lvs42HI7HRxs6nuZKvBZX3
To: <sip:[mobilephone]@sip.myprovider.com>
Contact: <sip:asterisk@[myip]:5060>
Call-ID: ctrsg83gzndDSt9le.4dmtoHVXeABVfW
CSeq: 1121 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 14.7.8
Content-Type: application/sdp
Content-Length:   233

v=0
o=- 861016884 861016884 IN IP4 192.168.1.3
s=Asterisk
c=IN IP4 [myip]
t=0 0
m=audio 28794 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (323 bytes) from UDP:[myprovider]:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [myip]:5060;received=[myip];branch=z9hG4bKPjTCmUWug.r6T.8eq.0d4NuM4QyegnESSy;rport=5060
From: "deskphone" <sip:[localnumber]@192.168.1.3>;tag=jHA.w0JCu.Lvs42HI7HRxs6nuZKvBZX3
To: <sip:[mobilephone]@sip.myprovider.com>
Call-ID: ctrsg83gzndDSt9le.4dmtoHVXeABVfW
CSeq: 1121 INVITE


<--- Received SIP response (354 bytes) from UDP:[myprovider]:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP [myip]:5060;received=[myip];branch=z9hG4bKPjTCmUWug.r6T.8eq.0d4NuM4QyegnESSy;rport=5060
From: "deskphone" <sip:[localnumber]@192.168.1.3>;tag=jHA.w0JCu.Lvs42HI7HRxs6nuZKvBZX3
To: <sip:[mobilephone]@sip.myprovider.com>;tag=aprqngfrt-t9ns203000126
Call-ID: ctrsg83gzndDSt9le.4dmtoHVXeABVfW
CSeq: 1121 INVITE
2

Provider need to tell you what are you missing or sending wrong in your INVITE request

3

You haven’t even got to trying to authenticate. The provider has rejected you without even challenging for authentication.

Is [localnumber] your account name with the provider, as some will identify you by your From: header user part.

1 Like
4

Inspired by David I took a look at the register packages:
the FROM header authenticates with [localnumber]@sip.myprovider.com, but the INVITE is sent with [localnumber]@192.168.1.3 (ip of my pbx).
How would I modify the domain in from? I am using asterisk 14.7.8, I am already using CALLERID(num) for the user part.
…
<— Transmitting SIP request (541 bytes) to UDP:[providerip]:5060 —>
REGISTER sip:sip.myprovider.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bKPjNXIeoAGva1VdTPMcRU.0xdZp5ckoOHvv
From: sip:[localnumber]@sip.myprovider.com;tag=ZVZfNL1sSCmy0ruebCbeMw1Qz8PIuqvs
To: sip:[localnumber]@sip.myprovider.com
Call-ID: uqmq54p272rQtfPIJOHjXpMfnK49vr8h
CSeq: 37115 REGISTER
Contact: sip:s@192.168.1.3:5060
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 14.7.8
Content-Length: 0

<— Received SIP response (467 bytes) from UDP:[providerip]:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:5060;received=[myip];branch=z9hG4bKPjNXIeoAGva1VdTPMcRU.0xdZp5ckoOHvv;rport=5060
From: sip:[localnumber]@sip.myprovider.com;tag=ZVZfNL1sSCmy0ruebCbeMw1Qz8PIuqvs
To: sip:[localnumber]@sip.myprovider.com;tag=9424e3d07702fd5bd45752fe654f1120.73cb
Call-ID: uqmq54p272rQtfPIJOHjXpMfnK49vr8h
CSeq: 37115 REGISTER
WWW-Authenticate: Digest realm=“myprovider.com”, nonce="[redacted]"
Content-Length: 0

<— Transmitting SIP request (722 bytes) to UDP:[providerip]:5060 —>
REGISTER sip:sip.myprovider.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bKPjNYve9ZAW8GrVEgEfZgXRWGI6Guu53jzl
From: sip:[localnumber]@sip.myprovider.com;tag=ZVZfNL1sSCmy0ruebCbeMw1Qz8PIuqvs
To: sip:[localnumber]@sip.myprovider.com
Call-ID: uqmq54p272rQtfPIJOHjXpMfnK49vr8h
CSeq: 37116 REGISTER
Contact: sip:s@192.168.1.3:5060
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 14.7.8
Authorization: Digest username="[localnumber]", realm=“myprovider.com”, nonce="[redacted]", uri=“sip:sip.myprovider.com:5060”, response="[redacted]"
Content-Length: 0

<— Received SIP response (418 bytes) from UDP:[providerip]:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;received=[myip];branch=z9hG4bKPjNYve9ZAW8GrVEgEfZgXRWGI6Guu53jzl;rport=5060
From: sip:[localnumber]@sip.myprovider.com;tag=ZVZfNL1sSCmy0ruebCbeMw1Qz8PIuqvs
To: sip:[localnumber]@sip.myprovider.com;tag=9424e3d07702fd5bd45752fe654f1120.d196
Call-ID: uqmq54p272rQtfPIJOHjXpMfnK49vr8h
CSeq: 37116 REGISTER
Contact: sip:s@192.168.1.3:5060;expires=120
Content-Length: 0
…

5

Sorry, I only know the answer to that for chan_sip.

6

@david551 I see you use chan_sip and dont give a try to pjsip, any particular reason that you can share,

7

There are historical reasons which I have gone into in a private message. The reasons don’t have any bearing on pjsip’s suitability for ordinary users, especially when compared with the official chan_sip.

1 Like
8

Thanks I 'm reading the message