[Freeswitch-users] [!!Mass Mail] Connecting Freeswitch and Asterisk with outgoingregistration

[Fred Schulz]

Do you have edited the acl.conf.xml in autload_configs to allow the Asterisk server?


2015-09-02 00:06:28.877729 [DEBUG] sofia.c:9001 IP 192.168.1.202 Rejected by acl "domains". Falling back to Digest auth.?


??


________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> im Auftrag von Markus Bönke <mbodbg at gmx.net>
Gesendet: Mittwoch, 2. September 2015 00:25
An: FreeSWITCH Users Help
Betreff: [!!Mass Mail][Freeswitch-users] Connecting Freeswitch and Asterisk with outgoingregistration

Hello,

I've  connected freeswitch with an asterisk server via sip trunk with the following configuration in my test environment:

Freeswitch side:

 <gateway name="sip.testprovider.com<http://sip.testprovider.com>">
              <param name="username" value="MyCustomSipTrunk1"/>
              <param name="password" value="easy123"/>
              <param name="extension" value="trunk"/>
              <param name="register" value="true"/>
              <param name="from_domain" value="sip.testprovider.com<http://sip.testprovider.com>"/>
 </gateway>


Asterisk side:

sip.conf

[MyCustomSipTrunk1]
type=peer
callerid="MyCustomSipTrunk1" <MyCustomSipTrunk1>
host=dynamic
nat=no
username=MyCustomSipTrunk1
fromdomain=sip.testprovider.com<http://sip.testprovider.com>
directmedia=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
secret=easy123
context=kamailio

extensions.conf

[kamailio]
exten => _X.,1,Dial(SIP/${EXTEN}@MyCustomSipTrunk1,60,tr)

If I send a call from asterisk to freeswitch, I can see the following in the log:

2015-09-02 00:06:28.877729 [DEBUG] sofia.c:9001 IP 192.168.1.202 Rejected by acl "domains". Falling back to Digest auth.
2015-09-02 00:06:28.877729 [WARNING] sofia_reg.c:2827 Can't find user [MyCustomSipTrunk1 at sip.testprovider.com<mailto:MyCustomSipTrunk1 at sip.testprovider.com>] from 192.168.1.202
You must define a domain called 'sip.testprovider.com<http://sip.testprovider.com>' in your directory and add a user with the id="MyCustomSipTrunk1" attribute
and you must configure your device to use the proper domain in it's authentication credentials.


If I create the user in the directory for the domain, it works - but why do I need to create this user however the gateway is already already registered and authenticated with the asterisk server?

Thanks

Markus



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150902/55c020dc/attachment-0001.html