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Messages - marian

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I noticed the calling party is behind NAT.
Maybe both yate and calling party are behind different NATs?

Check the audio on sip only:
Create the following rules in regexroute:
Call the numbers from sip.
For the first number you should hear a dial tone. This will check audio from yate to calling party.
The second rule will create a conference with echo: it will echo the sound from calling party back. This way you can check audio in both ways.

Make sure the tonegen and conference modules are loaded.


In the log we can see the calling party cancelled the call.
Can't see the timing.

Can you post a log with time? (-Dt command line option)

Sent/received jabber xml output would help also (jabberclient.conf, 'general' section print-xml=yes).

Linux / Re: audio problem on receiving telephone calls
« on: July 22, 2013, 02:24:51 AM »
There may be an issue with the incoming call audio notification.
You can disable the ringer in Settings -> Options. In the 'Telephony' page disable the Ringer checkbox.


Check if the openssl module is loaded.

If you built yate from svn install the openssl devel and re-build.

There is no microphone setup in settings.
Yate is using the default audio device.

Yate server for Google Voice / Re: Multiple GV Connections
« on: June 24, 2013, 01:12:50 AM »

Did you removed the line

Its purpose is to deny all unauthenticated calls: the sip module will not put the username parameter when routing an incoming, not authenticated, call.

Yate users hangout place / Re: Number selection in using Google Voice
« on: June 20, 2013, 07:59:32 AM »
If you are using Yate Client just press a keypad button.
The DTMF will be sent on the active call

Yate server for Google Voice / Re: Multiple GV Connections
« on: June 20, 2013, 01:38:19 AM »
When you set:
the rule in the left hand will match the called party number party.
If you set the same rule the first one will always match before the second one.

If you want to route an incoming call from a sip user registered to yate you may use the following rules:


    ${username}^USERNAME1$=goto GV_CALL;line=GoogleVoice1
    ${username}^USERNAME2$=goto GV_CALL;line=GoogleVoice2

; Set extra parameters for all calls to GV
; Call

If you want to do something else (e.g. other matching rules for calls) you should tell us what exactly do you want to do.

Yate server for Google Voice / Re: Multiple GV Connections
« on: June 19, 2013, 01:22:09 AM »

Your called party number matching rules are the same.
The first match will be always used.
That's why the calls go on the first account.

Windows / Re: no google voice calls
« on: June 04, 2013, 02:23:03 AM »

Do you have enough credit to call?

A log would help debug the issue (Check Debug Here and JABBER in the debug window).

Yate bugs / Re: Javascript route gwout
« on: May 29, 2013, 01:37:15 AM »
In regexroute you clear the caller and callername parameters when routing.
Do the same in javascript.

Yate users hangout place / Re: sorry for a really stupid question
« on: May 27, 2013, 02:52:28 AM »
modsuffix is by default '.yate'.
If you have another value for it then you re-defined it and re-built yate.

sslcontext set in ysipchan.conf listener must be the name of the section in openssl.conf describing the context

Hi, here is a configuration you need.
In regexroute and sip you should forward the SDP if are going to return the call to caller.
You might want to do it if the call is going to be sent to another destination also.
If you forward media yate won't handle it, it will be exchanged directly between call parties.




; Route sip call back to the source





Case 1: You have a SIP CNAM service provider.
You need sip_cnam_lnp module.
It must be explicitly loaded in yate.conf:


You must configure it:


Case 2: Use OpenCNAM
You need an script to make HTTP requests and handle the response.
It must implement a call.preroute message handler, make the request and put the calling party name in 'callername' parameter of the message.
I don't think you can use PHP scripts: it won't work on Windows.
You may use Python.

First you need the CNAM service provider.
1. You can use OpenCNAM, in which case you'll need the script you mentioned or write another one doing it
2. You can find a CNAM service you can query using SIP

You can handle calls on sip (from Obihai devices or any other source), make the CNAM query and return the calls back.

A minimal set of modules would be:

If you are going to use an external script to do CNAM you'll also need

For sip CNAM you'll need:

You'll need extra modules if you want support for Google Voice

When you'll decide what to use I can assist you with the configuration.

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