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Hi All, Just came across a weird issue just now. I'm no longer able to add/edit inbound/outbound destinations When I try to add, there's no textbox to enter the phone number: And when I try to edit, the phone number field is grayed out. Any idea that you can give me to correct this issue...

Hey, Thanks for this. Gave the information get this sorted. In essence this customer wants to present different outbound caller-IDs on the fly. So had to get this done in two places. 1. Dialplan -> Dialplan Manager -> Dialplan Name -> 2. Dialplan -> Outbound Routes -> Route Name (Copy of an...

Hi All, Need some help with another query. I've been scratching my head for two days on how to do this and my regex capability is abysmal. My requirement is to remove the 1st 3 digits of the destination number. For example, if i dial 6770412345678, then i want to set the destination number as...

Hi Folks, I've a situation where a forwarded destination needs to see the original caller ID. But they're only able to see the trunk/default caller ID. My setup is like this: Every domain has their own default caller ID set i.e the customer's main number under the dial plan variable There's...

Ha ha, you got me here. Exactly what I'm using. I'll post a full SIP trace for what's happening a bit later :)

Thanks for this. Yes, I'm using kamailio as a pass-through registrar & FusionPBX handles all registrations. The ultimate goal is for the proxy/SBC to handle all SIP/RTP communications and no extensions should connect to the FusionPBX directly. Hope this helps.

That is correct. This is just an example and using a directly connected extension. If the devices is behind Kamailio, the the IP's would be Kamailio's IPs. For example, it would do To: <sip:719@Kam.ail.io.IP:5060> I have the same issue with other proxies/SBCs as well due to this.

Hi Folks, I'm having a bit of an issue with the Invite headers sent to extensions. To put this simply, this for the last mile for inbound call i.e. when the call is being delivered to the handset. For example, X723 calls X719. This can be split into two parts. Part 1 : X723 -> FusionPBX (No...

Hi All, I've a situation that I need to address and would like your help in finding the right way forwards. Background: FusionPBX server has two NICs, one a Public IP, the other a Private IP. All SIP profiles are listening on the Public IP. Problem: I would like to create a SIP Profile which...

Hi Everyone, Is there a way for us to change the MOH based on the DID (example, FreePBX allows you to set the MOH at inbound DID level if we want to change default). The reason for this is that we have a few customers who changes the MOH for every holiday and at the moment, when they send this...

Hey All, I need some help on setting up DISA for a customer. They will have a dedicated DID which should ideally point straight to DISA so that they can toggle call diversion on/off. With FreePBX you can directly point the call to the DISA, but with FusionPBX I can't seem to find a way to get...

@Adrian Fretwell , Thanks for this snippet. Helped me to set things up quickly for this user :)

Hi Guys, Need some help with setting up speed dials for this one customer. He has about 50 odd numbers to be setup. What's easiest/fastest method of getting this done? Kind Regards

@JamesBorne , would you be able to share the XML part of this reg-ex? And also what's the order number you're using?

Hi Kenny Thanks for the update. I've already got the timezone setup as you've mentioned, but for some reason it doesn't work for this client with the time conditions. I could have easily done something like a manual offset and keep things simple, but their state observes daylight saving when we...

Hi All, Just wondering how you all cope with clients operating in multiple time-zones and daylight savings? Our server is set with AEST as the timezone, and we have a client on a different timezone. We've set the correct time zone for the domain and it is reflected on the CDR data, however, for...

Currently we have set the Advanced -> Devault Settings -> Voicemail -> storage_type to base64. This is currently causing issues with playback. It takes roughly 5 seconds for the greeting to play after the call has been answered by the mailbox. I'm thinking of switching this back to file...

Change the SIP Listen port to something other than the 5060. Most of the time this resolves the issue. If your WAN IP is static, then have EXT IP field filled in with that.

As the above script only moves yesterday's recordings, you can use the following commands to copy all the recordings to NFS. rsync -arvh --no-owner --no-group /var/lib/freeswitch/recordings/ /var/lib/freeswitch/storage/NFS/ psql --host=127.0.0.2 --username=fusionpbx -t -c "UPDATE...

This script is based off of "Convert yesterdays wavs to mp3" thread in this same section. The reason behind doing this way rather than changing the recording DIR to NFS share is that there's a probability the NFS share might go down. And when you have Lawyers and Financial people as your...