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The restore from freePBX will wipe out the incrediblePBX new module settings. So that is why I asked the questions what is the best way to get some of my old setting to the new one.

If you put in general discussion area, there maybe more people reading your post. Just my opinion.

sure! I really like to have ability to do the export/import from csv. At the moment, there is no easy way to get trunk, inbound out bound settings out except creating manually when doing new installation with new version of piaf. So I am for it.

So I have to recreate the IVR menu, extensions, trunks, inbound and outbound again manually. I know the extension I can export out and then input in. But IVR, trunks and in/out bound settings seem have no way to export and import.

As we know that we have new version of pbxinaflash from time to time. This time, the IncrediblePBX with google voice. The only issue I have is that we could not backup the settings we'v done with our older version and restore in the new version. How can we minimize the effort of recreating...

China does not block voip as far as I know. I have used softphone in hotel to connect to my piaf. I have good luck with zoiper. But I do have difficulties using x-lite. Another way to go is to bring an ATA, like Linksys Pap2. Many Chinese bring their ATA (offered by the company called italkbb)...

You are right. My mistake. I also put canreinvite=no in sip_general_custom.conf.

I don't understand why you say you do not have trunks. Even with a fxo card, you should have g0, g1, g2 trunk settings (I am using zaptel). You may not put anything there but you should have them identified in your trunk section. How do you dial out from one of your extensions to a specific pstn...

I added the canreinvite=no to my trunk peer details. Extra 0 in the cid is not a problem. Just deal with it on the redial. As I said, if extra 0 is only added to 010x., you can fix it on the trunk outgoing dial rule as follows: 0010|010+X.

My DTMF issue with cell phone is resolved by adding the canreinvite=no to trunk. check to see if that help you. I believe the PSTN line additional 0 is added by the ATA. You can either change the dialplan on the ATA or change it on the outbound route dialplan to removed the leading 0. Here is...

How can we get t38 support with pbxinaflash? What am I missing in settings? I got message like these below and the fax is not coming through. [2010-09-22 12:29:43] WARNING[10041] chan_sip.c: Unsupported SDP media type in offer: image 9158 udptl t38

I just added a new DID with T38 support. Try to test it and it works from outside lines. When I tried to call the line with my PBX other fax line, it failed. I was wondering how should I interpret the debugging message. "Unsupported SDP media type in offer: image 9798 udtl t38", is the...

I would disable the default trunks in the system since it has wrong information on userid and password. I just installed the new piaf and has no problem. Also check the email for the root to see if there is anything wrong reported.

Are you sure it is the router issue? Mybe it is your pbx trunk setting issue. If you are sure that is the firewall issue, the open the firewall.

Can anybody tell me if I have to manually create all settings after upgrade to the new version of 1.7.x and incrediblePBX?

Please, anybody can point me to the right direction?

So you could have incoming fax through sipgate and google voice but you would not have the outgoing fax since the outgoing fax has to go through google voice.

I still do not understand why this changing of context would fix your transfer.

After I build the latest incrediblePBX, I saw some errors on cron job. I need some one pointing me to the right direction to fix it. the email said: PHP Notice:, undefined variable: conf in /var/lib/asterisk/bin/module_admin on line 51 [FATAL] Cannot locate /admin/functions.inc.php. Any idea?

Does google voice support fax at all?