Configuration Linphone->Asterisk(FreePBX)->3G Dongle->external call

General Help
1

I have a fresh (well, almost fresh) installation of Asterisk, FreePBX on Mint 13 and Huawei E303S-2 dongle. Can someone guide me or point me to a step by step guide how to configure the system to be able to make phone calls from PC (Linphone) to external world via 3G dongle ?
Dongle is already installed, but I have no idea how, and where to configure system to redirect the call from Linphone to the dongle.

*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]

*CLI> dongle show devices
ID Group State RSSI Mode Submode Provider Name Model Firmware IMEI IMSI Number
dongle0 0 Free 8 0 0 POL E303S 21.158.13.00.641 80000000000 200000000000 Unknown

2

You could probably start with the wiki here, after reading your way through you will then know how to start configuring your system, consider it our version of a “step-by-step” guide for voip virgins.

3

Actually if he knows those commands and was able to install on Mint my guess is it’s just a language barrier.

I assume that you created an extension for Linphone and it registers to Asterisk OK?

All you have to do is create a custom trunk with a dial string of DONGLE0/$OUTNUM$

It looks like your dongle is configured and working. Once you have the trunk created simply create an outbound route for your local dial plan and point at the trunk you created. If you need any assistance on outbound routes see our documentation in the wiki.

4

Indeed Scott, once he has consumed the wiki his

sip show peers

will indeed return both the linphone and the Huawei and how effectively he defined them, currently not so much so.

5

sip show peers shows what is below.
I configured Linphone to listen on port 5061. I also have OpenBTS in the system, but I will take care of that later. I am facing a few problems here.

  1. It looks like SIM card registers on the network (dongle status - looks ok).
    But I do not know how to configure FreePBX, so when I put a tel nr ie. +49 176 8777 2222 in Linphone it will connect to dongle.
  2. “feature codes” in FreePBX by default are configured that “echo test” is on nr *43, however to dial that I have to type in Linphone “sip:*[email protected]:5060” otherwise *43 doesn’t work
  3. Where is an option in FreePBX that in Asterisk is called “autocreatepeers=yes” ? I intend to use this asterisk with OpenBTS and I need to be able to register SIM cards automaticaly.
  4. What is the right dial string “DONGLE0/$OUTNUM$” or “DONGLE/DONGLE0/$OUTNUM$” ?

*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
1000/1000 (Unspecified) D A 0 UNKNOWN
1001 (Unspecified) D A 0 UNKNOWN
2 sip peers [Monitored: 0 online, 2 offline Unmonitored: 0 online, 0 offline]

6

DONGLE/DONGLE0/$OUTNUM$

Autocreate peers does not have any function in FreePBX. If you are trying to create a switch for OpenBTS I think that FreePBX will get in the way more than help. You are far better off creating the simple dialplan you need directly in Asterisk Extensions.