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Playing with call recordings and when the record_ext is wav every few calls I playback sound like mickey mouse, audio is fast and indecipherable. when I change to gsm and download the files I don't think I have seen this issue. I would prefer wav files, but working is better. Anybody seen this?

regressions, priority over new features rather then fixing the broken features they have. Failing screens on GXP2130 that are only 2 years old. I have to test firmware before I roll them out for issues/regressions.

Uncle, I can't take the Grandstream firmwares any longer. Any good suppliers I should reach out to for good Yealink pricing? The supplier I currently get good pricing for on Grandstream does not carry Yealink.

you could still do what I have suggested, create a global rule and have a variable for the gateway. create a context specific outbound rule that sets ${Client_Gateway} sofia/gateway/${Client_Gateway}/1$1 I do this for CallerID

bad xml in the dialplan?

some calls that don't choose an IVR option are in the CDR some are not. I just made 4 calls from 2 phones. first call no VM, second call left a VM after not choosing a IVR option. All calls where in the CDR, so its random :)

They are in the public context and get transferred to my IVR, in the case of these calls the user never chooses and option (probably hoping to get transferred to reception) and the exit action of the IVR is going to VM. This is a cluster setup from the instruction on this site and fusionpbx 4.2

Sounds like the NAT is blocking the server. I don't use snom phones but the setting is probably related to this. http://wiki.snom.com/Settings/keepalive_interval

Sounds like a NAT setting, are the phone behind a NAT firewall? Are they running a current firmware?

I have gotten several calls over the last few days that are not showing up in the CDR log. anyone ever seen this?

If you use the same route for several clients could you not also match the context? ^(customer1.phonedomain.com|customer2.phonedomain.com)$ Then if you need to make a change, its as simple as editing 1 gateway for all the clients with the same route?

Is the Module loaded under Advanced->Modules?

This page explains how freeswitch config reloading works. https://wiki.freeswitch.org/wiki/Reloading Status->SIP Status has several reload options.

When you restart freeswitch

fixed by commenting out these lines in follow_me.php else { //set the outbound caller id number if the caller id number is a user //$dial_string...

I have my cell phone setup as a follow me for my extension. When you call my DID that goes straight to the extension the callerid on my cellphone is the callers as expected. when a did goes thru a ring group, the callerid generates the error in the callerid.

Are the phones registering with the server? looks like the call comes in and it try's to ring the extension but its not registered so it goes to VM.

I'm not getting caller id passed thru to cellphones with follow me feature enabled for extensions. This is in the call log for callerid, anyone see this before? caller_id "-ERR" <-ERR>

First, Thanks for the great easy to use script. Question 1, my older not clustered fusionpbx system is really out of date because the upgrade process was complex and I'm to behind to bother to try. With this cluster, whats the upgrade procedure? only the fusionpbx db syncs? I'm seeing this...