SIP Outbound Registration

Hello Everyone,
I’m looking for the solution too.
Got two @ in my username from my Provider.
When I try to register the provider VOIP server answer : 403 FORBIDEN Invalide user

Please Help

Question what are you using TrixBox, AsteriskNOW or Asterisk ?
if you are using TrixBox or AsteriskNOW you should ask on there forum
if you are using Asterisk you should change to chan_pjsip as chan_sip will be deprecated in October
there should be plenty of guids on how to setup registre on both TrixBox &AsteriskNOW’s forum
as your description for how to registre is more or less default behaviour

Hi egplpse can you post you config
and how your registre look like

This was the forum for AsteriskNOW, but it is long since dead, as should be this thread.

chan_sip already is deprecated. It is due for removal in October, but next year, not this year.

That shouldn’t be possible! This is an extract from the SIP RFC:

reserved    =  ";" / "/" / "?" / ":" / "@" / "&" / "=" / "+"
                     / "$" / ","
unreserved  =  alphanum / mark
mark        =  "-" / "_" / "." / "!" / "~" / "*" / "'"
                     / "(" / ")"
escaped     =  "%" HEXDIG HEXDIG
user             =  1*( unreserved / escaped / user-unreserved )
user-unreserved  =  "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"

@ is reserved, and is actually used to terminate the user part, and indicate its presence.

I’ve also never heard of an ITSP expecting more than one @, in the supposedly user part, and to make valid SIP. that needs to be escaped as %40.

Typically servers wanting @'s want you to put your user and domain in the SIP user field, but SIP also then requires a real @ and a domain.

As you should be using chan_pjsip, it allows you to specify exactly what goes over the wire, for registration, which normally should be a client URI in the form:


when they making this, confusing, requirement.

If they really require user@domain1@domain2, over the wire, they are badly broken.

Incidentally, suggesting actually sending the password in the clear, as the hijacked thread seemed to suggest, is an indication to find a provider who actually understands about security.

Hello @TheMark
Thank for your reply.
i run asterisk as my PBX.
When I use PJSIP as my sip channel i got the Uri tel from my provider and and message 416 unsupported Uri Scheme.
That’s why i’m actually use sip_chan

My register config:
register=> XXXXXXXX@provider_domain_name:secret@provider_ims_server_adresss:5060

The “+XXXXXXXX@provider_domain_name” is the username give by my provider.

But I got a forbiden from the provider server with 403 Forbidden “Authentication Failure”

Hello @david551
Ok got it.
But how when i try to escape the @ in the user part it is not translate in @. Iit goes clear like
register=> XXXXXXXX%40provider_domain_name:secret@provider_ims_server_adresss:5060

And i got the same error 403 Forbidden “Authentication Failure”
please help

Somebody can help me please ??

can i ask who your provider is
also can you get them to send a sample of how the sip REGISTRE must look like

Hello @TheMark
Orange Côte d’Ivoire is my provider.
All their sip account they provide are like : username: XXXXXXXXXX@Domain_name
and password

What I think was being asked for is the actual on the wire format. That way we can see if they require an invalid SIP URI or confirm that URI escaping should be used.

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