ahoman

Go to External Lines, click the phone line profile and go to Advance Line Settings, check the "Always replay audio (only use if one direction audio lost)"

Dial plan can help

I'm not quite sure Express Talk can do this. But X-Lite can. I have installed Axon. My X-Lite softphone can call xxx@domain.com without problem. You can refer to Express Talk HELP manual and check it.

Set one of the virtual interface "metric" to 1, and the other "metric" to 4 (any number greater than 1). The computer with Axon will always select "metric 1" as the first priority gateway. In Axon option->network, select "use stun servers to find the external IP address and port. not the static IP because you don't have one.

You may find the answer below: http://www.nch.com.au/kb/10064.html

Modify your dial plan, fill in the anyuser@example.com into "If sip host is" ... please refer HELP

The information in your message is not very detail. Anyway, if you are new to Axon and IVM, I suggest just install the Axon pbx first and create some extens for test. After that, integrate the IVM into Axon and try to create your own OGM. exten 199 is the default voice mail by Axon.

Sorry! From Hong Kong to Singapore needs IDD charges, I can't implement free service with additional cost unless I find partners or sponsors for such support. May be different area pbx systems willing to open to public and join together (form a world wide network) can find a solution and let all users enjoy free international call.

I would like to share my Axon pbx to public in order to test the traffic loading. At present I release 10 extensions for public and anyone can call Hong Kong landline. Please refer to the URL for details. http://www.ims.sytes.net Comments are welcome. Group the Axon pbx together and made the world wide speak FREE. Any idea?

You can combine the variables into one string such as.. $log_line=%date%.' '.%time% (and so on, the operator depends on what language you are using) then, use SQL comand INSERT INTO calllog (log_line) values ('$log_line') I also want to know other Axon variables besides of %date%, %time%....., I can not found them and I hope Axon can release these variables too.

Love to! HOW?

Create a sub-OGM , click change message audio and select clear the audio, set 0 to wait and repeat fields, in the then field select transfer call and fill in your mobile number . From your existing OGM, alter OGM--Key response ----6 ---dial 888 to OGM--Key response ----6 ---go to OGM and select your new sub-OGM. So when your press 6, the OGM will goto sub-OGM and then call your mobile.

My experience just for reference ONLY. I have a FXO gateway, I set up an external line to this FXO gateway and dial plan. Although Axon show fail to register but I still can dial an outgoing call to my landline. If I need Axon to accept incoming call from landline, I need to set the registration on FXO gateway to Axon as an extension . If anyone call from landline , FXO gateway will route to IVM and allow me to dial Axon extension follow the annoucement.

I know nothing about SPA..., but for a FXO device to hanup, it depneds: 1. a busy tone return from remote 2. detect a certain period of slience period. If you call from pstn, when you hand up, your side will return a busy tone to FXO, so FXO will on-hook. If SIP hang up first, it seems that Axon doesn't return busy tone to FXO such that SPA still off-hook.

If your phone and Axon server are located in the same lan, it should not has any problems. As you can ping the Axon server from phone, why not try to ping the phone from Axon server to make sure both sides can communicate well?

If you just want to make it works, then... open the following ports: 5060-5999 8000-65535 Because some softphone initiate the login at random ports, I use X-Lite and the login look like below (the red one is the login port used): sip:101@219.79.xxx.xxx:42712;rinstance=4917e3e329138985 expires at 02:57:26

Because no further information , let me guess... If your SIP phone (soft/hard) or gateway device with "silence detection" or something like that, disable it and "Voice Activity Detection" as well .

My example: Assume your Axon pbx server and the extensions are in different locations. At Axon server side , open the ports forwards to your Axon server 5060-5999 ( depends on how many extensions need to login) 8000-9999 At the extensions side, port forwards depends: 5060 to extension 1 5062 to extension 2 5064 to extension 3 and so on 8000-8999 forwards to all extensions For STUN server, also open port 3478 to all extension because different softphone handle STUN differently. Anyway, I suggest you setup a point to point VPN tunnel between server/extn sides as it is the simplest way, no port forwarding is needed.

I use VPN but no problem. May be you can check your VPN stability. I use L2tp VPN connection, PPTP can make VPN connection faster but drop off always, beside, don't set data encryption on.

My example: Install MySQL Install MySQL ODBC driver In MySQL create a database (example: pbx) Under pbx database, create a table (example: calllog) Fields: date--format date time--format time from_number--format VARCHAR/CHAR (30 characters may be enough) to_number-- same format as dfrom_number duration--format int Create a DSN in MySQL ODBC (DSN name example: MySQL) with database point to pbx database In Axon ODBC Log ODBC DSN: <enter your DSN name (my example is "MySQL")> User name: <your MySQL login name> Password: <your MySQL password> SQL COMMAND: <example INSERT INTO calllog (date, time, from_number, to_number, duration) values ('%date%', '%time%', '%from%','%to%','%duration%')> You can display the above information on web by means of php script. If you want to calculate the time for billing, set up another user table and set the tariff, then total cost=tariff X duration If your config is properly done, your can trace the SQL command under Axon log txt some what like this: 16:32:20 Executing SQL command: INSERT INTO calllog (date, time, from_number, to_number, duration) values ('2007-11-06', '16:31:55', '98xxx','sip:984201011@10.0.xxx.xxx:5060','24') Hope this can help

I want to set up the IVm to handle out going call (landline). That is : When any exten dial e.g. 111, the IVM will ask exten to input landline number, then call the landline number. I set up the OGm and after exten input the landline number, it will use "transfer call" function to call the landline via VoIP gateway. From the remote site exten, If I use VPN to connect to the Axon server (with IVM) and every thing is fine. But fail not under VPN mode, after I input the landline number, the IVM "transfer call" start and my softphone hand up. I have already opened port TCP 5000-65535 and UDP 5000-65535 on the remoet exten router's firewall. Any help? Anyone can use the followings to test my problem (English voice), after login, dial 109(OGM will take the call) for out going call SIP User ID or Number: 118 Password: irininej SIP Server: kcpbx.sytes.net Outbound Proxy: kcpbx.sytes.net Register Proxy: kcpbx.sytes.net Default Domain: kcpbx.sytes.net STUN Server: stun.xten.com STUN Server (Alt): stun.fwdnet.net