Hey everyone. MOH drives me insane not sure what I'm doing wrong. In the /var/lib/asterisk/moh folder I have the default ones and folders. After doing a bunch of changes and trying different file formats, I got it working by putting a hold WAV into the /moh/music-on-hold folder. The problem is I tried to change it, but uploading a new file there did nothing!
A community support forum for FreePBX users.
Our FreePBX version 18.104.22.168 has been running quite nicely for several months now, it is a small 5 extension system with 3 analogue trunks.
Today a user was cut off during a call, and had some other strange issues, a quick look at the system status showed the CPU at 100%. The log shows lots of the following messages.
[Dec 9 12:23:49] NOTICE chan_sip.c: Registration from '"4632" ' failed for 'somebody-elses-external-ip' - No matching peer found
I would like to ask if you guys have a documentation to start configuring the FreePBX that could use TDM400P, i can make calls extension to extension but when I place my call to pass FXO it doesnt make it.
I can't believe I couldn't find a post about this somewhere out there, but as I haven't (my own fault or otherwise) here goes:
I am running FPBX 22.214.171.124 on Asterisk 126.96.36.199 (64 bit).
For some ridiculous reason, I can't get extension CID to pass through the trunk- no matter what I do I end up getting the trunk CID. (The idea is to let specific people send out their DID number on CID... the trunk CID is the generic main line to the facility.)
I have PiaF 1.4 (Asterisk 188.8.131.52) with FreepPBX 184.108.40.206...
I have two extensions (201 and 205) which are connected with the Follow-Me option and have set the Recording Options to "On Demand" for both of them... But when i go to ARI i can see recordings for these extensions... These recordings are not for every call but they do exist for some of them... And i know for sure that the user didn't use the *1 In-Call recording... This also happens if i set Recording Options to "Never"... There shouldn't be any recordings for these extensions in ARI or in monitor folder...
With outbound routing in mind. I have a few providers and wanted to ensure that calls were distributed as evenly as possible between them. So no provider was getting the majority of calls.
Is there a way to do this at all?
The rationale for this is that I am about to install a 2 channel (and therefore 2 SIM GSM gateway. Each sim will have some free minutes so I want to ensure I use both sets of free minutes by sharing the outbound calls between them.
I hope I am making sense.
i have successfully installed FreePBX. now i want to create an IVR. i got the FreePBX in my localhost but the options 'IVR','Queues','Ring Groups' are not present. i could'nt predict the fault since i am very new to freepbx.i may be silly, somebody please help me, How to create IVR with in lan ....help me..
thanks in advance.
I am experiencing huge lag on conference call between software sip clients. I created conference which is used when we play games online (6 users connected). I am experiencing around 15 seconds delay (even if I don't play).
I also have SIP provider and make phone calls overseas every day. This is without any problem with absolutely clear phone calls.
My ADSL2+ line sync is 10000/1000. I think that I should not have a bandwidth problems.
I run asterisknow version with FreePBX (1.7.1) on Xen (Debian Stable) in HVM mode.
Running Elastix 1.6.12 and FreePBx 2.8
Symlink from modules failed
retrieve_conf failed to sym link:
/etc/asterisk/sip_notify.conf from core/etc
This can result in FATAL failures to your PBX. If the target file exists and not identical, the symlink will not occur and you should rename the target file to allow the automatic sym link to occur and remove this error, unless this is an intentional customization.
Added 16 minutes ago
We are not able to dial in or dial out but can dial internally.
I've been digging through some CDR bits and peices lately and a thought occured to me.
Could FreePBX insert the SIP CALLID values into the userfield of the CDR record?
This will help us a lot as we get our CDRs from our Trunk provider and they contain the providers callID. As far as I know there's no current way to match these with the CDRs from asterisk. However, if the SIP CALL ID of a call down a trunk was stored in the userfield column it would be a simple join in the databases.