Instances for communication between SIP and IAX Intranet and Internet and PSTN lines and mobile phones

At the request of the dynaguy brothers, I have attached a relatively complete experiment to colleagues who are still exploring:
(There are some problems with the 2.0beta experiment, so I will demonstrate it with the most stable 1.2.3)
In this test, we did not discuss the issue of using the SIP Trunk directly to connect other sip servers without board installation to enable the SIP users to make external calls.
This article is only used to discuss questions about using the fxo card to test internal and external line access.

I. experiment environment
1. trixbox server hardware
(1) Pentium iii500mhz/192 mb sdram/13G harddisk/BX motherboard With 4pci slot
(2) install an x100p fxo clone card in each of the two PCI Slots
2. Telephone lines and phones
(1) Two common PSTN Telephone lines are connected to line ports of two fxo cards.
(2) several other PSTN lines and analog phones (optional)
(3) IAX and SIP softphone software, such as X-lite and kiax
(4) two mobile phones (optional)
3. Broadband Network ADSL Environment
The Linux proxy and firewall server are connected to the Linksys Broadband Router over the Internet. The NIC is connected to the trixbox server. The topology is as follows:
Switch ----- Linux Nat server ----- Linksys ADSL Router ----- Internet ~ -~ -~ -~ -~ -~ PC with IAX/SIP softphone (remote)
|
| --------------- Trixbox server =================== 2 PSTN line ====
| --------------- PC with SIP softphone (local)
| ------------- PC with IAX softphone (local)
| ......

Ii. Installation Process
1. trixbox1.2.3.iso image files, burn CDs, start and press enter to automatically install, clear all original hard disk data
Never connect to other hard disks with data. In this way, data on all hard disks will be lost, even if you block other hard disks in the BIOS !!!

(If you don't have many hard disks to do experiments, for example, if you have installed a trixbox system on a Windows hard disk, press "expert" and press enter after the disc starts, the method for installing a custom partition is the same as the result of a carriage return. For trixbox systems, there is no difference except for retaining the original data and windows. But it takes longer. Export to/var/trixbox_load, and then run/var/trixbox_load/install_all.sh. In this manual mode, some machines are not lucky and may have to wait 1 to 2 hours for the Munin module to be installed, patience is required)
2. After the time zone and root password are defined, everything follows the default options. After the machine is restarted for the first time, the CD is taken out. After the second restart, the installation is completed and stops in login mode.

3. debug and configure Parameters
1. if you need to upgrade freepbx and clear the hidden Kernel panic error, you can refer to the post of dynaguy brother for the first step. My instance has not upgraded freepbx, but the machine will restart and crash, first, modify/etc/rc. d/rc6.d/k92zaptel solves hidden dangers!
2. Because two fxo cards are installed, modify/etc/zaptel. conf and add the following two blue statements:
Fxsks = 1
Fxsks = 2
Loadzone = us
Defaultzone = us
3. Modify/etc/asterisk/Zapata. conf and add the following three blue statements:
[Trunkgroups]
[Channels]
Busydetect = Yes
Language = EN
Context = from-zaptel
Signalling = fxs_ks
Usecallerid = Yes
Hidecallerid = No
Callwaiting = Yes
Usecallingpres = Yes
Callwaitingcallerid = Yes
Threewaycalling = Yes
Transfer = Yes
Cancallforward = Yes
Callreturn = Yes
Echocancel = Yes
Echocancelwhenbridged = No
Echotraining = 800
Rxgain = 0.0
TXT gain = 0.0
Group = 0
Channel = 1
Callgroup = 1
Pickupgroup = 1
Immediate = No
Faxdetect = incoming
Group = 1
Channel = 2
By now, the configuration file must be manually modified!

Iv. Call Test
The following operations are all completed by the freepbx graphical operation, which is very convenient:

1. Communication between PC phone numbers
(1) freepbx --> Tools --> module admin --> select all modules --> enable --> submit
(2) freepbx --> setup --> extensions add four extension numbers for PC Soft Phone testing (SIP Extension number 2201, 2202; iax2 extension number 2101, 2102)
(3) Configure Firewall port ing for SIP and IAX for Internet users
At this time, the internal and external network softlines can be normally called.

2. Other PSTN phones or mobile phones are called by the internal extension or the soft extension in any place, that is, the internal extension calls the external line:
(1) The system has set 9 as the outside dialing rule by default. If you only have one fxo card, the system does not need to adjust it. You can call an external phone or mobile phone.
For example, if you use X-Lite to log on to the system and register extension 2201, you can dial 9 first and then directly dial the external line number or mobile phone number.
Note: If the line you use to test on the fxo card is China Telecom wire connect, You need to dial two 9 lines and then dial the external line number. The first 9 is required by trixbox, and the second 9 is required by the call line of China Telecom. The principle of other types of lines is the same. In general, you only need to dial the first 9 for the home phone line test.
(2) I used two fxo cards to solve the call transfer test. At this time, we 'd better define the trunk sequence used by the outbound route.
First, modify the settings of the two zap trunks:
In freepbx, change the default trunk zap/G0 name to 1, that is, freepbx --> setup --> trunks --> trunk zap/G0 --> zap identifier (trunk name) the value of is changed from G0 to 1.
Then add a zap trunk and set the value of zap identifier (trunk name) to 2.
Finally, modify route 9_outside settings in outbound routes: Set the trunk Sequence Value to zap/2, that is, we fixed the call transfer (outbound call) by line 2/fxo card) when we call line 1/fxo card, if call transfer is set, trixbox will be able to transfer the call to line 1 from line 2!
At least till now, I still cannot transfer the phone number to line 1 from line 1. I personally think that the single fxo card of the PSTN signal cannot be completed!

3. Two PSTN phone numbers connected to trixbox are called by external PSTN lines or mobile phones.
At this time, the system will not handle the Call correctly. Do not believe you can use a soft phone to call 7777 or use a mobile phone to call the two lines of trixbox.

Here is the most exciting experiment:
Purpose: To test the external call function. The first step is to handle the call by the first-level IVR and provide a voice prompt (call XXXX. Please directly dial the extension number or dial 0 to serve you by the switchboard. Contact Mr. Zhang. Press 1. Contact Mr. Wang. Press 2. Test the agent queue by customer service. Press 3. Listen to background music. Press 4. Return. Press 5)
Press 0 to transfer incoming calls to a fixed phone or mobile phone;
Press 1, which is processed by the second-level IVR. A voice prompt is displayed (Press 1 for office phone number; press 2 for mobile phone number; press 0 for return)
Press 2, which is processed by the second-level IVR. A voice prompt is displayed (Press 1 for office phone number; press 2 for mobile phone number; press 0 for return)
Processing by 3 by the custom Quere
Press 4 to play a piece of music (when playing the music, press 0 to play again, press 1 to return a level-1 IVR)
Press 5 to return to the first-level IVR

Ladies and gentlemen, first prepare a wav-format voice prompt file. The simplest way is to use X-Lite to call * 77 and then call * 99 for testing, if you are satisfied, copy/tmp/unmamedivrrecording.wav and rename it and save it through SCP. In this way, you can quickly create a series of voice files for future use.

Procedure:
Add system recordings First: we strongly recommend that you directly use the upload with. WAV format method and name these recordings in the system, such as main IVR menu and Layer2 menu.

Then add IVR through digital predictionist
Note that enable direct dial is very important. You can call the extension number directly.

Create another queue for queuing testing, such as adding these extensions to the queue
2101
2102
2201
2202
2804
After submission, the system will automatically correct the format:
2101,0
2102,0
2,,0
2202,0
2804,0

The above operations also involve multi-layer IVR menu applications. The freepbx operation should be of no difficulty and I will not elaborate on it again. If necessary, I will try again.
Do not forget to set inbound route at last, and set all incoming calls to the first-level IVR, that is, the first-level welcome menu. In this way, the voice prompt is first heard after the external line is dial in.

Next let's look at the key problem: how to transfer an external call to another external call (or mobile phone?
There are two methods:
1. Create a queue and enter the target external line number to be transferred in the static agents file of the queue. Remember to add 9 in front and then put the queue in IVR for processing.
For example, add 9 to the front-end number and define it as queueqt. Then, use queueqt as the 0 action for welcome IVR.

2. create a user-less sip or IAX extension number, and set follow me for the number. In the extension list of follow me, enter 9 plus the outer line number and add # end, then, the Follow me project is provided to the IVR for processing.
For example, add "#" to the front-end number and define "followqt" as "followqt", and then use "followqt" as the 0 action for welcome IVR.

Is there a more direct and convenient way? I don't know.

(End)