[Spce-user] sipwsie as SBC

[Nicholas Papadakos]

Forgot to add , that I have added my PBX as sip peer in the sip peerings with its external address.



-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Nicholas Papadakos
Sent: Sunday, August 04, 2013 2:30 PM
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] sipwsie as SBC

Hello

First sorry aboyt the reply - I noticed itwent only to you after I send the email.
But I forwarded the msg to the mailing list anyway.



Now what I am trying to do is put sipwise act like SBC, meaning having it register to my asterisk that is behind nat ( sbc is in a datacenter) and have roaming users register on sipwise directly and be able to make and receive calls to and from the PBX(asterisk).


My problem is :

When I call from the PBX to the roaming user , registerd with sipwise I get got Authentication failed, no credentials in the kamailio logs.

The other way around works ok.

I tried putting in sip peer entry at the pbx the fromdomain=extern_ip_of_PBX but I got trivial results.
One extension is passing through , the other is still getting the same msg (Authentication failed, no credetials)

If I  disable force_outbound_calls_to_peer from the subscriber properties at sipwise everything works fine.

I also tried forcing externip=external_ip_of_PBX in sip.conf of the PBX but still no joy .

What is the correct logic for my scenario to work ?
I am a bit confused to be honest :)

Thank you in advance,

Nicholas Papadakos



-----Original Message-----
From: Jon Bonilla (Manwe) [mailto:jbonilla at sipwise.com]
Sent: Sunday, August 04, 2013 2:02 AM
To: spce-user at lists.sipwise.com
Cc: Nicholas Papadakos
Subject: Re: [Spce-user] sipwsie as SBC

El Sun, 4 Aug 2013 01:50:41 +0300
"Nicholas Papadakos" <panic at umbrela.org> escribió:

> Hello and thank you for your reply.
> 
> 

Please send your answers to the list


> I have the pbx listed as peer in the sip peerings server.
> 
> I didn’t notice before that asterisk is behind nat.
> I think it has something to do with the sip headers or something.
> 


The source ip of your sip signaling should be the ip address of your gw. In your logs AAAAAAAAA 


> My logs are like this :
> 
> 
> Aug  4 01:13:04 sip /usr/sbin/kamailio[32506]: INFO: <script>: New 
> request - M=INVITE
> R=sip:55102 at 62.X.XX.XX:5060;uuid=a82f6133-71f0-4115-a1ce-954dadc32107
> F=sip:2109317558 at 172.16.1.200
> T=sip:55102 at 62.X.XX.XX:5060;uuid=a82f6133-71f0-4115-a1ce-954dadc32107
> IP=AAAAAAAAA:5060 (127.0.0.1:5060)
> ID=155f7662310e8e462b554692697ed845 at 172.16.1.200 Aug  4 01:13:04 sip
> /usr/sbin/kamailio[32506]: INFO: <script>: Authentication failed, no 
> credentials -
> R=sip:55102 at 62.X.XX.XX:5060;uuid=a82f6133-71f0-4115-a1ce-954dadc32107
> ID=155f7662310e8e462b554692697ed845 at 172.16.1.200
> 
> 
> When I put fromdomain=62.X.XX.XX in the sip extension the call gets 
> through but Its stuck on a look to and from asterisk -sipwise. If I 
> disable force_outbound_calls_to_peer from the subscriber properties 
> everything works fine.
> 

Are we mixing things here? What's your scenario? Please describe it.



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