Philipp Hoffmann | 17 May 2013 22:12
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T.38 Transcoding

Hi,

I am trying to realize T.38 transcoding with Yate:

A  (SIP)--> Yate (SIP) --> Carrier (SIP) --> B (ISDN)

A supports T.38 (but no g711), the Carrier does *not* support T.38. That means, that Yate needs to transcode
T.38 to g711.

How can I configure this case in Yate?

We use YATE 2.2.0-1 (default debian package).

Regards,
Philipp.
Philipp Hoffmann | 15 May 2013 20:32
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German ISUP support

Hi,

is there currently any SS7 stack with German ISUP dialect available for Yate?

Thanks,
Philipp.

Paul Chitescu | 14 May 2013 11:43
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Re: Perform radius authentication lookups

Hi!

The Teles-UserDefined attributes are from a private dictionary and not decoded 
by Yate. Right now loading a dictionary is not supported, the RADIUS, Cisco, 
Microsoft and Quintum dictionaries are coded in tables at top of yradius.cpp

Also note that there's a level of indirection caused by passing the parameters 
through the authentication message. This matters for failed authorization.

If you want to return from a successful auth attempt:
  ret:h323-credit-amount=credit_amount
The attribute "h323-credit-amount" will be copied to parameter "credit_amount"

If you want to return from a failed auth attempt:
  ret-fail:Reply-Message=authfail_reject_text
The attribute "Reply-Message" will go to call.route parameter "reject_text"

Paul

On Tuesday 14 May 2013 12:03:42 pm Philipp Hoffmann wrote:
> Addition:
>
> [...]
>
> Our last issue is in set the returned radius attributes:
>
> --
> [nas]
> .
> .
(Continue reading)

Philipp Hoffmann | 11 May 2013 23:06
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Perform radius authentication lookups

Hi,

I am trying to perform radius authentication requests using yradius.conf:

The radius server expect two standard attributes, which have to been set up by Yate: The "User-Name"
(always the called party number, but without the first char, e. g. +49310 -> 49310) and the
"User-Password" (always clear-text: "radius").

For reference:

--
[nas]
add:User-Name=${caller} ;;;;; question: how can we cut the first char (+)? ;;;;;
add:User-Password=radius
--

The query goes through the radius server, but Yate do not provide the "User-Password" attribute.

Trace:

--
<yradius:NOTE> Using sections [nas] and [radius common] for authentication
<yradius:GOON> Ignoring unknown attribute of type 1
--

In the source code, the "User-Password" attribute is definied as binary, while "User-Name" is defined as
string. Is it possible, to set a binary (string) attribute in the yradius.conf file?

Thanks,
Philipp.
(Continue reading)

Paul Chitescu | 13 May 2013 18:08
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Re: Problem with Callfork and RTP/DTMF in progressing state

Hi!

What version of Yate do you run?

By default the DTMFs should be forked upstream - at least starting with Rev. 
4680 (2011-11-04). In Yate 4+ this should work.

Some changes in the SIP DTMF logic were made in Rev 5277 (2012-09-20)

Note that there are limitations on how a DTMF can be sent upstream in a non-
answered call. This may prevent sending them with fork or without.

The limitations are:
- SIP INFO needs the dialog to be established (so it doesn't work while 
ringing)
- RFC 2833 needs the upstream audio to be established and have some data (so 
it has clock source)
- inband needs the upstream audio to be established and a transcoder to the 
negotiated codec (and it may fail because of the codec)

You should raise the debug level of ysipchan to 8 or higher and check if you 
see messages like:

<sip/22:NOTE> Failed to send tones '6' methods=info,inband

You may see messages as the above only for some legs of a forked call. In the 
case above it was a Cisco phone than sends an 180 without SDP so no inband or 
RFC 2833 were possible at that stage. SIP INFO was not used because the dialog 
wasn't fully established (the call wasn't answered yet).

(Continue reading)

Diana Cionoiu | 10 May 2013 19:57
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Re: Re: Yate as MGCP Media Gateway

Hello,

The MGCP gateway in Yate is not full because it wasn't ment to be used. 
It was ment just for testing.
However we will be interested to develop more on that. Please drop me an 
e-mail at diana at null dot ro if you are interested in buying such a 
gateway.

Regards,
Diana

On 04/26/2013 03:01 AM, Eugene Prokopiev wrote:
> 2013/4/26 Eugene Prokopiev :
>
>> Is it possible to configure Yate as MGCP Media Gateway?
> I tried to connect Yate to Huawei SoftX with this minimal configuration:
>
> $ cat yate.conf
> [general]
> modload=off
> [modules]
> mgcpgw.yate=on
>
> $ cat mgcpgw.conf
> [engine]
> enabled=yes
> port=2427
> [ep yate]
> local_user=yate
> remote_host=10.10.10.100
(Continue reading)

Paul Chitescu | 13 May 2013 18:21
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Re: Maximum number of extensions in Yate

Hi!

Consider creating a tree of routes to limit the linear search length in each 
section.

You may do like this (based on 1st digit of number):

[default]
^[0-9]=goto start_\0
.*=echo Not starting with a digit?

[start_0]
...

[start_1]
...

...

This will reduce the complexity 10 times (if numbers are reasonably 
distributed).

Paul

On Saturday 04 May 2013 02:01:30 am ZZ Wave wrote:
> SIP only, G.711A, G.729
>
> Yate works in "normal" mode, with full signaling and RTP proxy
>
> regexroute is about 4k lines
(Continue reading)

Rubén Gracia | 30 Nov 2012 10:58

Re: SIP-H323 call limit

It was configured minport=16384 and maxport=32768. Nevertheless i 
increased it to 65536 with same result (yate crash with Segmentation 
fault when about 4096 calls are connected)
Any other idea?

Thanks for your help and regards,
Ruben Gracia

On 27/11/12 20:09, cc08 wrote:
> And how many rtp ports you give yate ?
> ...in rtpchan.conf
> ...minport, maxport
>
>
> Best regards,
> cc08
>

Ana Barbu | 27 Jan 2012 16:32
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What's the deal with SS7?

Hello,

Today I tried to find out if what is about to happen has anything to do 
with SS7 as I saw some work notes from Dana. She really hopes things 
will be ready soon, but wasn't quite sure if what she is doing is 
related to the big news.
What do you think? Maybe after the 2 days of thinking that follow, 
things will look up...

Have a great weekend!

Ana

Konstantin Reshetnikov | 27 Jan 2012 12:52
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RTPSession::incWrongSrc()

Hi, All !

Periodically, I see in yate log messages, like:

20120127103329.189866 <ALL> RTPSession::incWrongSrc() [0xad88af40]
20120127103329.210173 <ALL> RTPSession::incWrongSrc() [0xad88af40]
20120127103329.230451 <ALL> RTPSession::incWrongSrc() [0xad88af40]
20120127103329.250734 <ALL> RTPSession::incWrongSrc() [0xad88af40]
20120127103329.270926 <ALL> RTPSession::incWrongSrc() [0xad88af40]
20120127103329.291146 <ALL> RTPSession::incWrongSrc() [0xad88af40]
20120127103329.311361 <ALL> RTPSession::incWrongSrc() [0xad88af40]
20120127103329.331574 <ALL> RTPSession::incWrongSrc() [0xad88af40]

What is this ? Some problems with RTP flow ?

WBR,

Konstantin.

Ed Zietarski | 26 Jan 2012 22:13
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YATE as a SIP Proxy Server?

Dear Group Members,
 
I am new to YATE.
 
Can YATE be used as a SIP proxy server?
 
As an experiment, I would like to run two SIP clients and a SIP proxy server on my laptop (running Windows Vista).
 
That is, SIP client A calls SIP client B via the SIP proxy server, all on the same laptop.
 
If so, are there any example YATE configuration files or easy instructions to set up such a scenario?
 
I could not find anything specific on this in the YATE documentation.
 
Any help or advice would be appreciated.
 
Thank you in advance.
 
Ed Zietarski


Gmane