Messages

466

Windows / Re: yate calls dont go otrugh with spaces in the number

« on: November 15, 2013, 02:15:27 AM »

Hi,

The log shows yate sending the request 5 times and then giving up.
The server don't respond.

Checked with XLite 4.5: it removes spaces from called number and make the call without spaces.
Yate don't do that, it calls using the called number as entered.

467

Yate users hangout place / Re: Cisco Phone 7911 on Yate

« on: November 07, 2013, 07:51:32 AM »

Hi,

Freesentral does not support sip TCP.
You may add a feature request.

Can you add the yate log with UDP register from phone?
Log with message sniffer and maximum debug level.

468

Yate bugs / Re: Transcoding alaw to g729

« on: October 25, 2013, 01:23:33 AM »

cc08: the message is put by speex library used in speexcodec.cpp

469

Yate bugs / Re: Transcoding alaw to g729

« on: October 24, 2013, 08:13:12 AM »

To test it you need a calling application and a called application.
Just configure supported codecs to be different in both applications.
Yate will automatically transcode if caller/called party have different codecs.

470

Yate bugs / Re: Transcoding alaw to g729

« on: October 23, 2013, 12:58:55 AM »

When you are routing a call in regexroute it came from somewhere.
The application you are using to make the call is the calling party.

471

Yate bugs / Re: Transcoding alaw to g729

« on: October 22, 2013, 01:05:57 AM »

It is not easy to force transcoding.
Usually people want to avoid it.

Configure the calling party not to offer alaw.
Set the formats when routing:
.*=;formats=alaw

I suppose you do have a G729 data translator module for yate.
The transcoding won't work otherwise.


The incoming channel media won't change.
The outgoing channel format will be set to alaw.

472

Yate bugs / Re: Transcoding alaw to g729

« on: October 21, 2013, 01:23:39 AM »

What protocol are you using?
Why do you need to force transcoding?
Can you give us an example?

473

Yate users hangout place / Re: Billid in call.answered

« on: October 18, 2013, 01:31:14 AM »

Use chan.masquerade with message=call.execute and id=id_of_sending_channel

474

Yate users hangout place / Re: how to send 480 Temporarily not available when user ist offline using regfile ?

« on: October 18, 2013, 01:18:55 AM »

You can't do it using regfile.
You'll have to use regexroute.

475

Other Yate server issues / Re: Users authentication

« on: October 18, 2013, 01:14:49 AM »

The password you are returning in user.auth return value is not sent back on sip.
The sip module will check user credentials using the returned password and will accept or not the request.

476

Yate users hangout place / Re: yate to yate calling not working

« on: October 11, 2013, 12:55:46 AM »

Hi,

I can't help you without log file.

477

Yate users hangout place / Re: yate to yate calling not working

« on: October 09, 2013, 05:11:11 AM »

Hi,

Can you post a log?

yate-qt4 -CDat -vvvvv -l yate-qt4.log

478

Yate users hangout place / Re: yate to yate calling not working

« on: October 07, 2013, 02:34:45 AM »

Hi,

Can you describe your setup?
In yate client can you disable ringing? (Settings -> Options, Telephony page, uncheck 'Ringer')

479

Yate server for Google Voice / Re: Beginner in need of help!

« on: October 04, 2013, 04:09:40 AM »

Seems a sip timeout (maybe NAT issue).
The chan.hangup message shows rtp stats sent some packets, received nothing
It is possible the remote party didn't received any response to INVITE.
I have to see the sip signalling (don't forget the -Dt command line option).

To see sip messages:
Make sure the sip module debug level is at least 9 (put it at level 10 to see more):
yate-console.conf:
[debug]
sip=level 10

ysipchan.conf:
[general]
printmsg=yes

480

Other Yate server issues / Re: How to disable BYE after CANCEL

« on: September 30, 2013, 03:10:45 AM »

This one was added this year in march.
If you want to use it you have to update from svn.