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I'm so silly....sorry. Thanks a lot! I solved the problem!

I'm using last release of FusionPbx. I've a gateway registered in my pbx and when I receive a call this is what I see: Like you see the call is answered and fusion play an audio file. The call then should be forwarded to my internal context, instead happens something very strange! This is...

I already did that. I used this service online (http://g711.org/) to convert the audio file, enabled the mod_native_file but when I arrive to the moh fs says 60dad983-3f1d-460e-a35e-a6fb7d4c40be 2017-06-30 19:43:38.622832 [ERR] switch_core_io.c:859 Codec G.729 encoder error! 2017-06-30...

Sorry but I know in Asterisk you can provide different audio format (wav, gsm, g729). I guessed I could do that also in Fusiopbx in order to avoid FS has to transcode that. Is there a way to do that? Thanks

Hi, I'm using G729 passthrough in my FusionPBX. I've a hardware gateway supporting G729, hardware phone supporting G729. It works but I'm in trouble with MOH. Infact now music is not in G729 format and I can't upload g729 files using the interface. I should load g729 music so I would be able to...

Thanks for the clarification but I'm not talking about BLF. I'm speaking about what the phone calls "lines". Each line could be an extension of a different PBX for example.

Yes you got right. In my phone I can manage 4 SIP lines, so if I'm speaking I still can see others line blinking if someone calls. This works with Asterisk for example. Instead with Fusion+FS doesn't work. I receive immediatly a BUSY. If you say that this behaviour is normal, I'm fine. I was...

I just installed a fresh fusionpbx in a local machine with Debian 8. I was able to register 3 extensions: 2 on a phone Grandstream GXP2100 and 1 on another phone GXP1782. On the first phone I registered extension 100 and 102 on the first phone, extension 101 on second phone. I'm able to call...

Some more information hopefully are usefull. I was able to make work the gateway (patton) with inbound calls. Just a step backward: my pbx with FusionPbx is in the cloud so it has a public IP address (no NAT), my gateway (Patton 4114) instead is behind NAT in my office. I found interesting this...

Hi, is there some good soul that can give me some advice? Now I tried to use a Patton 4114 FXO but the problem I'm experiencing is exactly the same. I found this useful post http://lists.freeswitch.org/pipermail/freeswitch-users/2013-January/091703.html, I tried to do the same things but I've...

Works perfectly! Thanks @DigitalDaz !

Hi, I didn't find documentation about that. I've two Call center queues with different agents. Sometimes an agent that doesn't belong to the queue wants pick up that. Is there a way to do that? Thanks

Thanks, I did't know that FuxionPBX used php session. Thanks

Hi, how could I change the expire timeout for the web interface? Right now the timeout is too short for me. I don't see it in variables view. Thanks

Thank you so much! It works perfectly!

Thanks a lot. It worked. I'm still struggling with a problem when calls come in though. Internal calls now use G729 codec, calls that are coming from Voip provider instead arrive in PCMA (I'm sure the provider offert G729). Infact then the call arrive I see: dce8a2d4-c0ea-4e92-9011-4a9846dc1401...

Hi, I'm taking a try of mod_bcg729 (https://github.com/xadhoom/mod_bcg729). I was able to load the module: type,name,ikey codec,ADPCM (IMA),mod_spandsp codec,AMR,mod_amr codec,BroadVoice16 (BV16),mod_bv codec,BroadVoice32 (BV32),mod_bv codec,G.711 alaw,CORE_PCM_MODULE codec,G.711...

I made some progress. According to some post I found, I created an extension in fusionpbx for my ht-503 (not a gateway). So now I'm able to receive calls. I'm not still able to make calls because I'm trying to understand if I should make calls using this particular extension. Seems a bit weird...

I posted all the log I saw. Maybe I missed something? According to your suggestion I changed the configuration of my HT-503 and requests are sent to port 5080. The sure thing is that I see arriving the call in Freeswitch but it is rejected. I guess I need something that in Asterisk is the...

Someone has some hint for me? Thanks