PIAF installed version - 2.0.6.3 under VMWARE
FreePBX version - 2.11.0.2
Running Asterisk version - 11.1.0
Asterisk Source version - 11.1.0
Dahdi source version - 2.6.1+2.6.1
Libpri source version - 1.4.12
OS - Centos release 6.3 (Final)
Kernel version - 2.6.32-279-14.1.e16.i686-32bit
Incredible version 11.1.0
Ports forwarded - 5060-5082 (UDP), 5222 (TCP), 10000-20000(UDP)
Issue 1.
Inside my LAN, I can hear audio from the Csipsimple app IF STUN server is UNCHECKED. The opposite happens when I’m out of my LAN and am on my cell data plan.
In order for CSipSimple to work:
LAN - uncheck the STUN Server box
3G - check the STUN Server box.
Csipsimple registers, I can make and receive calls, but if I don’t enable/disable the STUN setting respectively, I get no audio from either side.
Anyone else having this problem?
Issue 2.
I have an obi110 that I’d like to make as part of the freepbx setup. What I want to do is have my Google Voice # ring both, Csipsimple and the obi100. Just the same, I’d like to be able to make calls out from either device and have it show up the same Google Voice as the caller id. Is this possible and can you point me in the correct direction?
Thanks!
Issue #1.
When in the LAN, you may not need to set the STUN support because afterall you have forwarded your ports and thus are visible to the sip proxy.
When using 3G, you are in effect passing through another network which may or may not be forwarding the necessary ports to make a connection, as such, the need for the STUN service.
Issue #2.
For incoming calls, create a ring group for those two devices. Route incoming calls from GV to that ring group. In that ring group, you can not set whether all extensions ring simultaneously, one at time, etc.
For outgoing calls, set a unique prefix or a dial prefix for the GV trunk dial plan. Prepend called numbers with that prefix. e.g. calling 1-888-888-8888, set a prefix, say 101 on the GV dial plan. You will now use 101-1-888-888-8888 on your extensions to dial.
Thank you for the replies.
I’m curious how other voip providers like callcentric does it when I connect to their sip with csipsimple from the same droid and not having to enable and disable stun each time I connect and disconnect from my LAN. If I placed my freepbx in the dmz, this essentially leaves it wide open, ports and all on the internet, correct? I’ll give this a quick try but I thought I had the same results as though it was behind my router.
As for issue 2, that’s going to be a deal breaker. As I was hopping that it would be more seamless like how Google voice rings all phones enabled on the account.
I think he meant to say can, instead of “cannot”. Have you looked at the ring group functionality?
Asterisk only provides marginal support for stun. You have to make your SIP and RTP ports exposed to Internet if you want to be able to access them remotely.
Asking Asterisk to terminate the call and turn it around to another SIP trunk is a complex order for a device behind any kind of network translation. Asterisk is a B2BUA not a proxy.
Asterisk is all new to me. I will read up further on ring group functionality.
I have forwarded the mentioned ports, 5060-5082, 10000-20000, and 5222. Are there others required to negate the need of stun servers or is it the nature of VoIP over the Internet?
I will need to read up more about the use and need of stun servers in regard to asterisk and sip soft phones outside of the LAN.
Thank you for your replies.
You don’t need 5222 and anything but 5060 and RTP range. I suggest you go to FreePBX SIP settings module and reduce the range of ports. You need 2 per call leg.
sorry it should have read “can now set” rather than “can not set”… my apologies.
No worries.
I was able to resolve my Issue 1 and possibly an easier solution to Issue 2 after some searching on the CSipsimple site.
Csipsimple allows the disabling of using the STUN Server setting within the SIP Account that you create for registering to the PBX.
What I did:
- Enable STUN by default
- Create 2 SIP entries with identical credentials to the PBX. Label one LAN, other WAN.
- On the LAN SIP, make sure you’re in Expert mode or else the STUN settings are not visible. Change the STUN for SIP to DISABLE.
Now when you’re inside your own LAN, deactivate the WAN account and activate the LAN. Vice versa when you’re off your LAN. This works great.
For those that don’t want to even have to manually change the accounts back and forth, using Tasker and/or Locales android app to sense a change in LAN or Cell Data mode and have the change done automatically. I have not done this yet but that’s what I’ve found is a possible solution.
My issue with this current setup is the caller on the other side hears an echo of themselves. I don’t hear any echos on my end at all. The other is the audio on my end sounds like I’m in a concert hall. I’ll have do more searching.
Alternatively, since I have an Obi110 and the call quality is superb on the obi with no echos at all with lots of options available in regard to linking the mesh of obi, freepbx and csipsimple. I’ll give them a try.
Possible scenarios:
Obi <> Csipsimple
Obi <> Freepbx <> Csipsimple
Freepbx <> Obi and Csipsimple
If anyone have any of the above setups already completed with good results on sound and care to share the setup instructions, I’d appreciate it very much.