Dial tone/ringing on incoming calls

[d.kli...@gmail.com]

We are busy implementing our new phone server.

It's working fine (xmpp, calling and voicemail), but we have one problem left.

When someone is calling to us, that person doesn't hear any ringing. The call does arrive at our phone and when picking it up we can hear each other.

How can I enable the ringing (early media?) for incoming calls?

Thank you in advance.

[d.kli...@gmail.com]

I will give some more information on this (rephrase it):

We have a sip trunk without username and password.
we have not entered any username and password and disabled the register on startup.

What happens now with the currect provider is that they do not "send or trigger" the ringback from sipXecs. version 14.04.20141017110701.

So we get called the internal phones to ring but the caller does not hear anything until the call is picked up by the user or the IVR.

We find in old (version 4) documentation a referance about send ring back over IP in the SIP menu but that is not available in the 14.04.x version. We can not find anything in the SBC and Gateway settings.

The SIP trunk provider say's that the end point (in oure case the SBC) should send the ringback not the Sip trunk provider.

Please can somebody help out on this? Everything else is working like a charm!

[Jimmy]

Who is the provider?

[d.kli...@gmail.com]

The provider is Prionet. We have already contacted them, but they tell us it is something we need to configure.

Op donderdag 2 juli 2015 20:47:51 UTC+2 schreef Jimmy:

[Mihai Costache]

Hi,

Can you please provide more information regarding your environment?

Are you using internal SBC (Sipxbridge) or an external mediagateway/sbc (who is the manufacturer)?

If this is audiocodes then you have an option called Play Ringback Tone to Tel under SIP General Parameters

If this is a sangoma device see:
 ftp://ftp.sangoma.com/vega/legacy/1-step-by-step-configuration/Base%20documents/Initial%20config%20-%20R6%20Vega%20100%20E1%20%28SIP%29_06.pdf
Chapter 14.4

Where are the inbound calls finished ? You are calling in Autoattendant or directly an internal extension? What type of phones are you using? This phones support 100rel?

A pcap will also help.

And why are you using 14.04? In 15.04+ version there is a new feature under System--> Voicemail-->"Transfer By Bridging the call"  ---> if you enable this then calls through freeswitch will play ringback tone

Thanks!

Best Regards,

Mihai Costache

"No problem can withstand the assault of sustained thinking. "

                                -  Voltaire

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[d.kli...@gmail.com]

We are using the Sipxbridge.

Inbound calls were first going to a hunt group, we also tried straight to a phone. For now we are using a workaround which makes inbound calls go to an autoattendant which will then forward the call towards a hunt group. This way the caller gets to hear some waiting music.

We are currently using two Siemens Optipoint 420 standard.

I will see if I can provide a PCAP after the weekend.



Op vrijdag 3 juli 2015 15:15:10 UTC+2 schreef Mihai Costache:

[George Niculae]

If calls going to autoattendant then you have to enable following option in System > Voicemail

Transfer By Bridging the call

true false	(Default: false)
	Bridge call for transfer through IVR / AutoAttendant (default transfer method use SIP REFER method) Enable this option to fix integration with ITSPs / SBCs that does not support SIP REFER and if you want to have ringback played while transferring call.

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[George Niculae]

Hm, I misunderstood, you are using this option only as an workaround...

[Mihai Costache]

Hey,

In this case you will probably need to use an external SBC.

You will find  a freeswitch SBC example here - you will need to add also the ringback option in fs dialplan:
http://wiki.sipxcom.org/display/sipXcom/FreeSWITCH+SIP+Trunking+Gateway
https://wiki.freeswitch.org/wiki/Early_Media

If you are in the mood to test with freeswitch then please update wiki page with your success story here:
http://wiki.sipxcom.org/display/sipXcom/Custom+FreeSWITCH+programming

Best Regards,

Mihai Costache

"No problem can withstand the assault of sustained thinking. "

                                -  Voltaire

To view this discussion on the web visit https://groups.google.com/d/msgid/sipxcom-users/c63500a2-0efc-40ea-81c1-0fb026279965%40googlegroups.com.

[Michael Picher]

Usually in this instance the provider is not routing into 5060 UDP.

Mike

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[Michael Picher]

Sorry, 5080udp

[d.kli...@gmail.com]

In the end the provider has activated early media. So the problem is solved now.

Op woensdag 1 juli 2015 15:48:07 UTC+2 schreef d.kli...@gmail.com:

[djg...@gmail.com]

Thanks Mihai,

Even 4 years later this was exactly what I needed for a 3CX system with Sangoma Vega 100G GW.

i.e. used your ftp link which still works. Went to chapter 14.4

At command line issued:

Set _advanced.isdn.user_progress=1

Now incoming callers hear ringing tone when calling a 3CX that has a Vega 100G PRI gateway in front of it.

Is Google and the Internet a wonderful thing !

Thanks again for taking the time to answer this years ago.

Darren

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