[Freeswitch-users] Skypiax load error

Thu Nov 5 07:19:37 PST 2009

Hi All,

I once meet the Skypiax load error issue and some guys infomed me that there is no configuration file for Skypiax. 

When I follow these intructions ->
http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances

I still have some difficulties unstanding this:

." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to Debug\conf\autoload_configs."
I did not find such directories in my freeswitch folder. Did not understand what "src" means, I checked the freeswitch folder and did not find such a folder named"src". There is a folder named"mod" under freeswitch while look into "mod" folder there are only some DLL files and can not find endpoints and etc.

2.You'll probably build the "Debug" version

 I just build this from the precompiled binaries and then launched FS. I'm not sure what I launched is in debug mode or not.

If anyone can offer some help that really be appriciated.

Thanks

Daniel Zeng
From: freeswitch-users-request at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 41, Issue 40
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 5 Nov 2009 06:57:44 -0800

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--附转发的邮件--
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 5 Nov 2009 08:13:46 -0600
Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media

I did not ask you to send me a ladder diagram.
I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me)

1) start FreeSWITCH
2) run the cli command: console loglevel debug

3) run the cli command: sofia profile internal siptrace on
4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog)

Also please answer brian's question.  What phones and/or sip devices are involved in this call.

On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana <hjqlopez at hotmail.com> wrote:

Thanks for your time,

-The scenario is still the same:

Always bypass media.

Environment 100% NAT free :-)

Call established from A to B through FS. Then... 

Blind transfer from B to C (Refer-to: C)

RTP should go directly between A and C.

-With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio.  Only RTP from C to A (due to the lack of reINVITE to A, after C answers).

Please check SIP diagram here:

http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html

-What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-)

http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html

-You could check the sofia debug for r15332 here:

http://pastebin.com/m6f2b3836

Best regards,

Humberto

>

> I don't know what you are talking about anymore.

>

> The scenario I had tested is when a call is bridged in bypass_media=true

> bridge

> and you blind transfer that call back to the dialplan

>

> as soon as it hits the routing state it will resume media.

>

>

> it has been confirmed to not work and confirmed to have been fixed several

> time and if you are still having a problem you must have something blocking

> some of your packets or something .

>

> You have to understand that sip is a protocol and your description is

> completely non-standard.

> Perhaps you should get a console trace and attach it to a jira. The trace

> probably makes more sense to me.

>

> sofia profile internal siptrace on

> console loglevel debug

>

> reproduce and attach the whole capture.

>

>

>

> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote:

>

>>

>> Hi,

>>

>> I tried r15332 and set in the sofia profile:

>>

>> a) bypass_media_after_bridge=true only

>> b) bypass_media_after_bridge=true, param name="media-option"

>> value="resume-media-on-hold"/>

>>

>>

>> In both cases FS is hanging up the initial call (A to FS) after accepting

>> the REFER to C:

>>

>> A <- reINVITE with FS' SDP <- FS

>> A -> 200 -> FS

>> A <- ACK <- FS

>> A <- BYE <- FS

>>

>> The call to C is not even tried.

>>

>> I found this line is the logs that could give some idea:

>>

>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup

>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]

>> after sending the ACK for the reINVITE

>>

>>

>> Regards,

>>

>>

>> Humberto

>>

>>>please try r15326

>>>I think i have it working.

>>>

>>>I recommend for optimal results you set bypass_media_after_bridge=true

>>>either as a global or in your DP in place of bypass_media=true

>>>

>>>

>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana

>> hotmail.com>wrote:

>>>

>>>> Hi Mike,

>>>>

>>>> I re-tried with trunk rev 15319 but I got almost the same behavior:

>> There

>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted.

>> But

>>>> still there is no reINVITE for A (with C's SDP) after the call from FS

>> to C

>>>> is established.

>>>>

>>>> Anyway, we decided for now to do a different implementation but if you

>> want

>>>> to explore more in this issue count me in ;-)

>>>>

>>>>

>>>> Thank you very much!

>>>>

>>>> Humberto

>>

>>

>> _________________________________________________________________

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>>

>

>

>

> --

> Anthony Minessale II

>

> _________________________________________________________________

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--附转发的邮件--
From: rob4manhere at gmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 5 Nov 2009 08:52:05 -0600
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

Hi UK,

 From what I've done and read, the caller-controls (in  
conference.conf.xml) can be modified to almost anything you can think  
of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a  
caller control just for those with the moderator flag.  So unless you  
want everyone able to mute/kick everyone then you can't do it.

The wiki seems to indicate this as well:

"Be aware that the caller-controls are applied across the entire  
conference. You cannot enter one member of the conference using caller- 
controls ABC and then enter a second member using caller-controls XYZ."

http://wiki.freeswitch.org/wiki/Mod_conference

I think this might be a limitation of mod_conference.  Perhaps one of  
the pros can chime in if I'm off-base or there's some nifty way to  
accomplish this.

Cheers,
Rob

On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote:

> Any ideas on the below...has anyone implemented the below:
>
> Once I have the Moderator and Participants logged on, how do I  
> invoke the moderator previlidges, LIk esay muting everyone/someone  
> or kicking someone out of the Conf and the like?
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ujjval Karihaloo
> Sent: Monday, November 02, 2009 12:52 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> Rob:
>
>   Once I have the Moderator and Participants logged on, how do I  
> invoke the moderator previlidges, LIk esay muting everyone/someone  
> or kicking someone out of the Conf and the like?
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman
> Sent: Friday, October 30, 2009 9:34 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> Hm, strange.  I haven't seen that before.  Can you pastebin your logs
> at debug level?
>
> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
>
>> It's strange... a tcpdump tells me that there is no DTMF from my
>> provider when using IVR, but when I call into a TN that goes
>> directly into the Conference App, I see DTMF from the provider.
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>> ] On Behalf Of Rob Forman
>> Sent: Friday, October 30, 2009 7:23 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>>
>> I've never had any problem with that.  Is your logging at debug level
>> so you can see the RECV DTFM in the log/fs_cli?  Are you calling from
>> a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider
>> isn't passing them through.
>>
>> Make sure your logging is turned up then try something simpler, like
>> calling the echo application, and see if DTFM comes through.
>>
>> Rob
>>
>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
>>
>>> Rob:
>>>
>>> For some reason, I don't see the DTMF appear on the fs_CLI when
>>> using the below configuration....so it basically timesout.
>>>
>>> UK
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>> ] On Behalf Of Ujjval Karihaloo
>>> Sent: Monday, October 26, 2009 9:21 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>>>
>>> Thx a lot Rob, reading the wiki your way or using IVR seems  
>>> correct..
>>> ===============
>>> The wiki also says that the wait-mod might be  "used in conjunction
>>> with an IVR where the moderators are authenticated with an extra
>>> pass-
>>> code", which is what I did.  I guess that's why I didn't understand
>>> the point of the +pin.
>>> ======================
>>>
>>> I will try it out.
>>>
>>> Again thx a lot for your help. Will keep everyone posted.
>>>
>>> ________________________________________
>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org
>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com]
>>> Sent: Friday, October 23, 2009 12:22 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>>>
>>> I just re-tested with the pin in my dial plan:
>>>
>>> <action application="conference" data="conference 123456 at default
>>> +flags{}+1234" />
>>>
>>> And it doesn't challenge me for the pin.  I just drop right in.  I
>>> figured this is how it was intended, since the wiki says the pin is
>>> set initially and only challenged in later attempts [by future
>>> callers]:
>>>
>>> "The first time a conference name (confname) is used, it will be
>>> created on demand, and the pin will be set to what ever is specified
>>> at that time: the pin in the data string if specified, or if not,  
>>> the
>>> "pin" setting in the conference profile, and if that is also
>>> unspecified, then there is no pin protection. Any later attempt to
>>> join the conference must specify the same pin number, if one existed
>>> when it was created. "
>>>
>>>
>>> The wiki also says that the wait-mod might be  "used in conjunction
>>> with an IVR where the moderators are authenticated with an extra
>>> pass-
>>> code", which is what I did.  I guess that's why I didn't understand
>>> the point of the +pin.
>>>
>>> I'm sure there's a scenario where its used and useful, the wiki just
>>> doesn't explain it.
>>>
>>> Rob
>>>
>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote:
>>>
>>>> Well first off you're not defining a pine here...
>>>>
>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference
>>>> pin
>>>> number]
>>>>
>>>> That might be why its not asking for a pin.
>>>>
>>>> /b
>>>>
>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>>>>
>>>>>   <entry action="menu-exec-app" digits="1" param="conference
>>>>> 123456 at default+flags{}" />
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
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>>>> http://www.freeswitch.org
>>>
>>>
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>>
>>
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>
>
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--附转发的邮件--
From: rob4manhere at gmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 5 Nov 2009 08:57:08 -0600
Subject: [Freeswitch-users] Wideband / HD phones

Hey all,

Looking at buying some high def phones.  Any recommendations  
(preferably based on experience) for hardware based on product  
quality, standards compliance, features integration with Freeswitch,  
etc?

Thank you!
Rob Forman

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