Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Brian West
http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios Please review Scenario 1 the bold text. ;) /b On Jun 21, 2008, at 10:56 AM, Ivan C Myrvold wrote: > I put this into internal.xml, and this seems to do the trick: > > > > > Ivan > > Den 21. juni. 2008 kl. 15:47 skrev Ivan C

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Brian West
Let me guess.. you have phones behind the same NAT talking to this profile also? If that is the case you'll need to create a new profile with ext-sip- ip and ext-rtp-ip set for the outside world to talk to.. then take those settings off the profile your phones behind the nat talk to freeswi

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Ivan C Myrvold
I put this into internal.xml, and this seems to do the trick: Ivan Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold: > Yeah, the call arrives nicely now, but the audio is only 1-way. > I have port forwarded all the ports in my router to FreeSwitch, but > still experiences just onew

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Ivan C Myrvold
Yeah, the call arrives nicely now, but the audio is only 1-way. I have port forwarded all the ports in my router to FreeSwitch, but still experiences just oneway audio, from FreeSwitch to the DID caller. Looks like the RTP is not forwarded correctly to FreeSwitch. Could it be my Linksys router, w

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Good to know it snapped into place now! :P /b On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote: > This is all making sense to me now. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
Ah, I put the "apply-inbound-acl" in the wrong XML file. I put it in the external.xml. When I instead put it in the internal.xml, I got it working. That is of course the correct place, because it is bound to port 5060, and the inbound from Voxbone comes in on port 5060, as you correctly guess

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Let me guess they only send to port 5060? If you have your ACL's setup correctly those two IP's will be let in without auth. If you have the profile that runs on 5060 on your FreeSWITCH box with auth-calls=true Add this to that profile This should allow them thru without auth. If y

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
I do not have outbound registation to Voxbone, because Voxbone is only incoming. I am not registrating Voxbone at all. In their FAQ, they have how to configure for Asterisk: [81.201.82.20] host = 81.201.82.20 type = friend insecure = very context = your-context canreinvite=no [81.201.82.21] ho

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Ivan, Do you have your outbound registration to voxbone on the default/ internal profile? If so then your acl's might be wrong. The one sure fire way to do this is to just setup another profile without auth on a different port and run with that. /b On Jun 20, 2008, at 3:20 AM, Ivan

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
As I understand the ACL, it only controls which external machines with the IP address range given in the acl.conf.xml are allowed into FreeSwitch. So this doesn't help me much with the DID, as FreeSwitch sends a "Proxy Authentication Required" SIP message back to the Voxbone server. How can I

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-18 Thread Ivan C Myrvold
I added to my acl.conf.xml this http://pastebin.freeswitch.org/4640 and this to external.xml When I start freeswitch, I see that the settings are used: 2008-06-18 09:11:09 [CONSOLE] switch_core.c:769 switch_load_network_lists() Created ip list voxbone default (deny) 2008-06-18 09:11:09 [

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-17 Thread Brian West
You'll need to setup an ACL to let them in without authentication. Look at the Asterisk to FreeSWITCH section on the wiki. http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk /b On Jun 17, 2008, at 12:31 AM, Ivan C Myrvold wrote: > I have used Freeswitch with a DID from Voxbon

[Freeswitch-users] Can I get SIP DID working?

2008-06-16 Thread Ivan C Myrvold
I have used Freeswitch with a DID from Voxbone working on IAX, and that have been working so well. But now Voxbone will discontinue the IAX service, so I have to get DID working on SIP, and I am wondering if that will work at all in my configuration. Freeswitch is behind nat, so when I get a