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Today's Topics:
1. Re: mod_skinny --> Calling numbers with various lenght (MS-R-IT)
2. Re: Delayed sound after answer (Anthony Minessale)
3. Re: Delayed sound after answer (Oleg Stolyar)
4. Re: [ERR] sofia_reg.c:2110 "some gateway" Registration Failed
with status Unauthorized [401]. failure #37 (Snabel Kaabiya)
(Brian West)
---------- Forwarded message ----------
From: "MS-R-IT" <martin.schoepfer@ms-r-it.de>
To: "'FreeSWITCH Users Help'" <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 8 Apr 2014 19:58:15 +0200
Subject: Re: [Freeswitch-users] mod_skinny --> Calling numbers with various lenght
Hello Nathan,
thanks for your reply can i have checked this hack but i don´t think this is
a solution to communicate. Are there any plans to get this function into
freeswitch in the future?
I check now how I could handle the cisco phones temporary with a small
asterisk chan_sccp connected to freeswitch. Is there any way to get
chan_sccp transformed to
Freeswitch! Because its Opensource too! I only able to program some smalls
things in C or C++ so I have to ask if it is any reason and then I can check
it by myself!
Thank you for your informations. I think you have luck with NANP!! :-)
Mit freundlichen Grüßen
Martin Schöpfer
____________________________________________________________________________
__________
MS-R-IT
Martin Schöpfer
Regelungs- und Informationstechnik
Bergham 3
83052 Bruckmühl
Tel.: 08062 7238870
Fax.: 08062 7238871
Mobil: 0176 80088253
-----Ursprüngliche Nachricht-----
Von: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Nathan
Neulinger
Gesendet: Dienstag, 8. April 2014 19:41
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_skinny --> Calling numbers with various
lenght
On 04/08/2014 12:10 PM, MS-R-IT wrote:
> Hello,
>
> yes they can overlap you can have different length with overlap like
> my company siemens the central has the number 3-digit prefix (yes this can
also be) then 4-digit four our site (like 9221) and 0 for our central and
5910 for me directly.
>
> There are many else expressions I could give you emergency call in
> Germany is 110 and 112 but there can be number with 110XXXXX too!
This is not something you'll be able to do with the current built-in dial
handling for mod_skinny currently because of how it processes the dialing.
The key part of the current processing for this is in
skinny_handle_keypad_button_message() where it calls
skinny_session_process_dest() with the non-null 2nd to last argument
(append_dest).
The processing sets call to routing state after each digit in that routine.
What would need to happen is some sort of time delayed callback to have it
not send the call to routing state until a certain time has elapsed.
You could do a workaround of requiring a terminating digit (# or something
like that) to "dial" - that would work with
current code, but is definitely a hack.
-- Nathan
------------------------------------------------------------
Nathan Neulinger nneul@mst.edu
Missouri S&T Information Technology (573) 612-1412
System Administrator - Architect
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---------- Forwarded message ----------
From: Anthony Minessale <anthony.minessale@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 8 Apr 2014 13:04:48 -0500
Subject: Re: [Freeswitch-users] Delayed sound after answerIts probably based on your network. Are you behind NAT? Sometimes it takes a few seconds for the automatic translation to kick in.You can add a sleep 1000 after answer to ensure you have established media before you play the tone.On Mon, Apr 7, 2014 at 5:10 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,I have the following in my dialplan:<condition field="${<some field>}" expression="<some expression>"><action application="answer"/><action application="set" data="api_hangup_hook=curl <some URL>"/><action application="playback" data="tone_stream://v=0;%(1000,0,600)" /><action application="conference" data="$1@default"/></condition>Notice the line in bold. In most cases the tone plays fine but in some cases it is shorter than 1 second and in a few cases it does not play at all. Has anyone run into this? Is there a way to make the occasional media delay after answer go away?
_________________________________________________________________________
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--Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
ClueCon Weekly Development Call
---------- Forwarded message ----------
From: Oleg Stolyar <olegstolyar@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 8 Apr 2014 12:12:42 -0600
Subject: Re: [Freeswitch-users] Delayed sound after answerThanks Anthony, I'll try that.On Tue, Apr 8, 2014 at 12:04 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Its probably based on your network. Are you behind NAT? Sometimes it takes a few seconds for the automatic translation to kick in.You can add a sleep 1000 after answer to ensure you have established media before you play the tone.On Mon, Apr 7, 2014 at 5:10 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
_________________________________________________________________________Hi guys,I have the following in my dialplan:<condition field="${<some field>}" expression="<some expression>"><action application="answer"/><action application="set" data="api_hangup_hook=curl <some URL>"/><action application="playback" data="tone_stream://v=0;%(1000,0,600)" /><action application="conference" data="$1@default"/></condition>Notice the line in bold. In most cases the tone plays fine but in some cases it is shorter than 1 second and in a few cases it does not play at all. Has anyone run into this? Is there a way to make the occasional media delay after answer go away?
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
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--Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
ClueCon Weekly Development Call
_________________________________________________________________________
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---------- Forwarded message ----------
From: Brian West <brian@freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 8 Apr 2014 13:25:05 -0500
Subject: Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status Unauthorized [401]. failure #37 (Snabel Kaabiya)
This is a bug in the provider, If they want to start auth over they should say stale=‘true’, this way the NC will start back at 1. not 18e
--
Brian West
brian@freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX 2531
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On Apr 8, 2014, at 7:07 AM, Steven Ayre <steveayre@gmail.com> wrote:
> It's not the first request because it's sending a challenge response using a nonce, which it must have received in a previous challenge (401)... It can't just make that up.
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