PJSIP Error No Auth Credentials For Realm(s) ‘asterisk’ In Challenge

Home » Asterisk Users » PJSIP Error No Auth Credentials For Realm(s) ‘asterisk’ In Challenge
Asterisk Users 3 Comments

Hi all,

we are trying to move our servers from chan_sip to chan_pjsip. At this time no problems with phones, they all register fine and can place calls. But for a trunk we face problem and can’t place calls despite the fact that registration is OK. What we get is:

[2018-04-16 16:08:33] WARNING[18665]:
res_pjsip_outbound_authenticator_digest.c:178
digest_create_request_with_auth_from_old: Endpoint: ‘sip.xxx.tld’:
Unable to create request with auth. No auth credentials for realm(s)
‘asterisk’ in challenge.

Our setup:

[sip.xxx.tld]
type = registration retry_interval = 20
max_retries = 0
contact_user =
expiration = 3600
transport = transport-udp outbound_auth = sip.xxx.tld client_uri = sip:@sip.xxx.tld server_uri = sip:sip.xxx.tld

[sip.xxx.tld]
type = auth password = username =

[sip.xxx.tld]
type = identify endpoint = sip.xxx.tld match =

[sip.xxx.tld]
type = endpoint context = from-xxx.tld aors = sip.xxx.tld deny = 0.0.0.0/0.0.0.0
permit = /32
permit = /32
dtmf_mode = rfc4733
disallow = all allow = alaw,ulaw,g729

[sip.xxx.tld]
type = aor contact = sip::5060

Registry:
========
zone-s*CLI> pjsip list registrations

 
 
==========================================================================================

 sip.xxx.tld/sip:sip.xxx.tld sip.xxx.tld Registered

Objects found: 1

PJSIP is listening on port 12345, chan_sip on port 5060. The peer end is a Kamailio 3.3.4 if it matter.

What could be the problem? Does anyone have a PJSIP asterisk registered against Kamailio?

Thanks for any hints

Daniel

—

3 thoughts on - PJSIP Error No Auth Credentials For Realm(s) ‘asterisk’ In Challenge

  • The remote side challenged for authentication but your endpoint has no “outbound_auth” configured, so chan_pjsip has no idea of how to authenticate.

  • Le 16/04/2018 à 16:52, Joshua Colp a écrit :

    Thanks Joshua, that did it. We already tested a sort of by inserting line = yes and endpoint = sip.xxx.tld in registration stanza but this didn’t work

    Again, many thanks.

    Daniel

    —

  • Outbound registration and outbound calling have no relation. The “line” option is strictly for INBOUND calls from what you have registered to.

    —
    Joshua Colp Digium, Inc. | Senior Software Developer
    445 Jan Davis Drive NW – Huntsville, AL 35806 – US
    Check us out at: http://www.digium.com & http://www.asterisk.org

    —