Thanks, this was a step in the right direction :-) Outbound dialing is working but I don't have a ringtone during dialing and when I answer the phone I don't hear anything. This might be due to the NAT'ed SPA3102. 2011-04-14 23:35:53.986767 [DEBUG] sofia_glue.c:2971 AUDIO RTP [sofia/sipinterface_1/sip:trunk@192.168.1.153:5063] xxx.xxx.xxx.xxx port 25902 -> 192.168.1.153 port 16426 codec: 0 ms: 20 Any idea how to fix that? On 12 April 2011 22:55, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
So, it's a user (from FS point of view).. A gateway (from somewhere) registers to your FS instance. Ypu want to dial to this gateway, just dial to it as a normal user. But just try to override the invite uri;
{sip_invite_to_uri=<sip:REAL_NUMBER@${domain_name}>}user/registered_user@${domain_name}
I just wrote this from my head, so it might have errors, but the big idea is to dial the user, but at the same time override the sip invite uri, to a different number.
At least this is what I do in this situation.. :)
/Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Nick Rosier [nick.rosier@gmail.com] Skickat: den 12 april 2011 20:26 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Gateway with dynamic IP address
The problem is this isn't a user, it's a gateway that registers itself as a user (so FS does not register to it but the other way around). I need to dial to it as a user and specify the number it has to dial.
On 12 April 2011 12:20, Steven Ayre <steveayre@gmail.com> wrote:
To dial a user you use <action application="bridge" data="user/username"/>, FS then figures out the Sofia URI for you from the registration.
-Steve
On 11 April 2011 20:53, Nick Rosier <nick.rosier@gmail.com> wrote:
On 5 April 2011 22:45, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
What you wan't to do is to add a user. Then you dial this user, which by then is registered in FreeSWITCH, and it will find the path.
So no gateway in this case, it's when you want to register to an external server, a user is when someone registers to you, and you wan't to be able to dial outside through this.
/Peter
Can someone help me with the URI. It's driving me crazy. This is what I've got but it's not working:
<action application="bridge" data="sofia/sipinterface_1/trunk1@pbx.domain.com/$1"/>
What am I doing wrong?
N.
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