Dear.
Excuse some off topic of Wikipbx, but today i solved my problem; wikipbx and FS
working now.
Best regard.
--- On Sun, 5/24/09, Jason White wrote:
> From: Jason White
> Subject: Re: [Freeswitch-users] FS & Wikipbx Help
> To: freeswitch-users@lists.freeswitch.org
> Date: Sunday, May 24,
Hi Guys,
I have install Freeswitch with version : FreeSWITCH Version 1.0.4pre7
(13238M) Started.
I load the openzap module after install the wanpipe modul and everything
running , after I query the status of the Sangoma Card (A104D quad E1)
we got error like this :
2009-05-25 11:07:45 [ERR] swit
And the tag line could include something about big savings.
On Sun, May 24, 2009 at 7:30 AM, Brian West wrote:
> I vote for Moose Penis Telecom
>
> /b
>
> On May 23, 2009, at 3:31 PM, Diego Viola wrote:
>
>> Thanks everyone, we just choose our own name :).
>>
>> On Fri, May 22, 2009 at 1:57 PM, D
Congrats..So what did you decide on, this is your chance for a free plug ;)
On Sun, May 24, 2009 at 6:31 AM, Diego Viola wrote:
> Thanks everyone, we just choose our own name :).
>
> On Fri, May 22, 2009 at 1:57 PM, Darin Weeks wrote:
>>
>> I just went through naming my own thing -- pretty h
Hi Brian and Anthony,
We need to move back a couple of steps here. I fully understand the A
leg cannot enable SRTP unless it sends descriptors in the original
INVITE. As the A party works as expected lets not discuss that any
further as it clouds the waters so to speak.
What I am trying to achi
On Mon, May 25, 2009 at 8:58 AM, mashudi wrote:
Anybody could share the profile configuration for softphone register
> behind NAT?
http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA
http://wiki.freeswitch.org/wiki/NAT
http://wiki.freeswitch.org/wiki/NAT_Traversal
http://wiki.freeswitch.org/wi
Hi Guys,
I implement FS as SIP Proxy including media (sip & media) , the server
sitting behind NAT and the softphone client sitting behind NAT, so the
configuration would be :
voip provider (172.17.67.91) (172.17.67.12) FS on VPN network
(172.17.67.13) - NAT (222.192.12.23) public NAT
*Here is a better FreeSwitch IVR starter guide..*
http://wiki.freeswitch.org/wiki/Getting_Started_Guide
http://wiki.freeswitch.org/wiki/IVR
http://docs.freeswitch.org/group__switch__ivr__menu.html
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
http://wiki.freeswitch.org/wiki/Freeswitch_I
On Sun, May 24, 2009 at 8:17 PM, bakko wrote:
> I just installing FS in my new server. When i can make tests y will write
> you.
bakko:
its really quite simple ... and should work straight out of the box
first install your server -
http://wiki.freeswitch.org/wiki/Installation_Guide
then get
FERNANDO VILLARROEL wrote:
> My install of wikipbx was succesfully, but i have problem for registering a
> Softphone Xlite for testing; i look the next warning in FS_CLI:
>
> freeswi...@internal> 2009-04-28 06:13:58 [WARNING] sofia_reg.c:1701
> sofia_reg_parse_auth() Can't find user [...@192.16
With apologies for the incorrect address in the header, if you reply to this
follow-up instead of the original message we should be fine for the remainder
of the thread.
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http://list
Hi, Niall Crosby,
may it could help :
http://www.freeswitch.de/xwiki/bin/viewrev/OpenZAP/WebHome?rev=8.1
I have Sangoma A104D quad E1 , I follow that link, and it's work.
Niall Crosby wrote:
> Hi List,
>
> Am about to buy a Sangoma A101 Single Port T1/E1/J1 to plug our
> Freeswitch box into an E1
I've already discussed this with a few members of the community, but I would
like to raise it with a wider FreeSWITCH audience.
Since upgrading to libtool 2.2.6a (now the default in debian testing), I can't
successfully link mod_portaudio.so. The system is Debian Testing, x86_64
architecture.
ldd
Hi List,
Am about to buy a Sangoma A101 Single Port T1/E1/J1 to plug our
Freeswitch box into an E1 at the telco.
Is there any reference on how to set up Freeswitch to interface with this?
thanks,
Niall.
--
--
Sremium Ltd.
Reg Number: 451937
Mobile: +353 (0)87 2393174
Web: www.sremium.com
Here is a start:
http://wiki.freeswitch.org/wiki/IVR
http://docs.freeswitch.org/group__switch__ivr__menu.html
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate
http://wiki.freeswitch.org/wiki/Mod_commands
Best regards,
-E
Gpro.ws
i need a manual sort of thing to help out for implementing a basic IVR using
freeswitch
--- On Sun, 5/24/09, freeswitch-users-requ...@lists.freeswitch.org
wrote:
From: freeswitch-users-requ...@lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 35, Issue 116
To: freeswitch-users@lists
Hello Fernando,
The best you can use is this one:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide
very easy ;)
David
On Sun, May 24, 2009 at 5:35 AM, FERNANDO VILLARROEL
wrote:
>
> Dear All,
>
> I am new user of Freeswitch and searching information i find wikipbx:
>
> http://www.wiki
I just installing FS in my new server. When i can make tests y will write
you.
Regards
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I would really appreciate this.
Some of the problems I have been unable to solve are:
* inbound callerid presentation. SIP inbound to nokia have "0123456"
(quotes included) instead of just a number that would then be matched
with a phone book entry to display the name of the caller.
* random call
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