Modify Contact In PJsip

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Asterisk Users 6 Comments

Hi Guys

We are using the wizard to configure our pjsip trunk(see below)

How do we get this setting to work

contact_user=username

We want to change the contact field in the sip invite to display the username of the trunk

[trunk_defaults](!)

type = wizard

transport = transport-udp

endpoint/allow_subscribe = no

endpoint/allow = !all,g729

aor/qualify_frequency = 30

registration/expiration = 1800

contact_pattern=xxx

[xxx](trunk_defaults)

sends_auth = yes

sends_registrations = yes

endpoint/context = extensions

remote_hosts = xxx.xx.xx.xx

accepts_registrations = no

endpoint/send_rpid = yes

endpoint/send_pai = yes

outbound_auth/username = xxx

outbound_auth/password = xxx

contact_pattern=xxx

6 thoughts on - Modify Contact In PJsip

  • The Contact header can not currently be modified on a per-endpoint basis and takes its values from the generated From header. On a global scale it could be controlled using the default_user global option. Otherwise there’s no real way without adding explicit support for it.

  • Hi Joshua If i put the default_user option per endpoint would it work?¬†
    So what exactly does the contact_user option do?
    I know that in freeswitch there is the option extension-in-contact.We  basically need to achieve the same functionality 
    Thanks

    ——– Original message ——–
    From: Joshua Colp
    Date: 2015/10/19 13:03 (GMT+02:00)
    Subject: Re: [asterisk-users] Modify Contact in PJsip

    The Contact header can not currently be modified on a per-endpoint basis and takes its values from the generated From header. On a global scale it could be controlled using the default_user global option. Otherwise there’s no real way without adding explicit support for it.

  • No, it’s a global only option.

    It sets the Contact user in an outbound registration so that the URI
    dialed by the remote SIP server may contain that user (or may not, depending on their configuration/deployment).

    It would require modifying the code and adding support.

  • Do you know if this can be achieved with the standard sip stack in asterisk?

    Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)

    Switchboard: +27 (0)10

  • If you are referring to chan_sip I don’t believe so but it is possible there is some obscure option or method to do it that I am aware of.

  • Ok thanks Joshua

    Do you know what this error means when I dial out in pjsip and the call fails

    Unable to create request with auth.No auth credent als for any realms in challenge

    Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)

    Switchboard: +27 (0)10