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Yes Amekassa you need to resample the audio file from MP3 to a wav file - 8k 8bit mono wave file in u/a law

You need to flag it on the incoming call to the handset so the handset's choice of ring tone is honoured. eg you have a phonebook --> contacts with the number added to it, you can add a picture, the...

Martin, why go to the "cloud" as it is just someone else's "computer" appliance.. why not look at the QX500 with 228 sip extensions (base 100 + additional 128 Sip licence) with 80 concurrent calls -...

You would need to write your own custom scenario to achieve this without delays. It can be a mammoth task so be mindful that whilst it can be done, it takes time. I had one customer in an inbound...

call routing table needs to have an allow rule to dial the numbers you are wanting them to dial externally. This does pose a security risk though.

Source Number Pattern/Caller ID Modification ...

Sounds like you do not have enough line keys on the telephone to process the attended part of the call... add another line key on the telephone and make sure that it can handle 2 x calls and that...

I always would resample the MOH file to a .wav 8k 8 bit mono using the likes of WavePad Sound Editor (we use the paid version ) - look for it on NCH's website. The application comes in handy for all...

Stop using epygi stun and use the external ip, reduce the stun timer to 1 hour

the wan port of the quadro is usually set to 100 mb half duplex or 10 mb half or full duplex. If you have an internet connection that is better than 10/10 mb up and down, you will need to place a...

This is purely a guess, however - look in the telephones to see if they are honouring it in the first place, check that the provider is actually sending the CLI - look under sipua for the call. Epygi...

using my method you then change the delay for no answer timer to set it to another destination under the suplemental settings

Use a virtual group and have the telephones assigned for that group with associated ring type - ie sequential, ring all etc...

"Route all incoming SIP calls to Call Routing" is enabled

Why are you using that setting?

You need to look at the sipua logs

http://www.valcom.com/pdf/v2003a-e_V2003ahf-e.pdf

Page 7 has the input and output settings

If you enable the LAN port and add a phone to that port does the same behaviour occur or does it go away for sip trunk to that extension call?

Just a suspicion I have about the media negotiation...

If you have a back up try a factory reset, the password will then be restored to admin and 19. HTTP requests from the lan side will work, unless that units firmware is in error, then you might have...

Change the codec support on the extension to be in line with the codecs used on the Epygi to the Bria software - extensions management -- extension codec to the right of the extension.

Clear the cache of your browser or use Internet Explorer / Mozilla and clear the cache before going to the web of the Epygi.

Some handsets overcome this by automatically changing the UDP SIP port it listens and transmits on, other handsets you have to change the port manually, in all instances your local router needs to be...

Set up forward of the virtual extension - supplemental services to the vm of the extension.

Dont use pattern matching.

Call Queue Size: 4
Max Calls Presented to Extension: 1 <------ 4
Enable Redirection on Timeout: ticked
Call Queue Message Repetition Count: 5
Call Type: Auto
Address: CRT <----- 52

Source...

Hi Ed

Just 1 x suggestion is needed to be enacted upon first.

In the extension routing table, create an extension 52... forward that immediately to the call routing table as (Auto) that will...

Set 52 to be forwarded to the call routing tables and allow anything to dial 52 in the CRT to the mobile number. Unless you have RTP that is not lining up then the call should go through with audio.

Try using OTHER as the handset type, and make sure that the extensions codec preference doesn't override the handsets PCMA / PCMU depending on the country that you are in.