Bear in mind that FS will accept both 2833 and INFO in any profile on an
inbound call. Param "dtmf-type" is valid only for outbound calls from the
profile.
Ognjen
On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine
wrote:
> Hello,
>
> I have Polycom phones which send only RFC-2833 (or inband which
Hey Hadley,
jump up on irc sometimes.
Regards,
Ognjen
On Mon, Nov 9, 2009 at 9:26 PM, Hadley Rich wrote:
> On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote:
> > Get an ATA with a Dect handset it works much better... the Snom M3 and
> > the Aastra are one in the same and they both do not liv
Add the following:
.
after
in local extensions default example, or change it globally previously than
this extension. You can join us on IRC if you can any more questions
(sekil).
Regards,
Ognjen
On Tue, Nov 10, 2009 at 4:01 PM, Piotr Żurek wrote:
> Hello.
>
> Thank You developers for
Hello,
I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS.
Phones are registering to CCME and FS simultaneously. So far, everything is
working just fine.
I think SCCP will be obsolete in the future since even Cisco is working more
and more on SIP. OTOH, I really hate SIP images
Polycom responds to SIP server dhcp option (I think it's 120). I haven't
still seen any other phone doing that.
Ognjen
On Wed, Sep 23, 2009 at 6:16 PM, digilord wrote:
> Brian,
>Is there code someplace that I can get that will help with the
> Polycom
> DHCPINFORM way?
>
> Thanks
>
> On
Hello Eric,
I just interconnected FS/openzap and Panasonic via E1 PRI trunk (euroisdn)
using libpri stack, and it works just fine.
Feel free to drop on irc for any help.
Regards,
Ognjen
On Wed, Sep 2, 2009 at 7:29 PM, Michael Collins wrote:
>
>
> On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond
I currently use FS for a managed PBX services for many companies. It's so
configurable and extensible, I don't think I would change it for anything. I
have Cisco CME and SCCP phones in my office all interconnected to FS doing
logic and thinking. I also have Asterisks as PSTN PRI gateways. FS does a
Hello,
I authored that wiki article. The following key will work:
fnc=blf+sd+cp;sub=1...@$proxy
You need to make sure that presence is not off in the profile. Also "cp" in
the key will enable you to do the intercept of "ringing" call to watched
extension. For further help please join #freeswitch
CRIBE, MEASSAGE, OPTIONS)
>* The number of messages to handle was not that much (some thousand
> subs).
>
> For my understanding this should also be possible with Freeswitch with
> bypass_media. Right?
>
> Best regards
> Peter
>
>
>
> Ognjen Seslija schrieb:
&
I vote for viotel.
Regards,
Ognjen
On Fri, May 22, 2009 at 6:26 AM, Diego Viola wrote:
> Hey guys,
>
> I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
> cool names for my VoIP company, if you know some please tell me :)
>
> Diego
>
> _
Hello,
FS by design is B2BUA, and it cannot route INVITEs and other SIP methods. It
can however, bridge a-leg to b-leg with or w/o media and doing plenty other
cool stuff much better than commercial projects. I suggest joining us on irc
to detail your setup so we can help you.
Regards,
Ognjen (se
t; I also installed the Nokia Configuration Tool (V3).
>
> Which settings did you apply for getting the crypto line?
>
> So far I only got TLS to work, there is no crypto line so far.
>
> Best regards
> Peter
>
> Ognjen Seslija schrieb:
> > Hi,
> >
> >
Hi,
after installing Nokia Configuration Tool I managed to get E61i to offer
srtp only it sends crypto line in the RTP/AVP which is not rfc behaviour:
2009-04-27 12:11:12 [ERR] sofia_glue.c:2785 sofia_glue_negotiate_sdp()
a=crypto in RTP/AVP, refer to rfc3711
Can anything be done with this?
Ogn
To share my experience: I had issues with echo with many E1 trunks in
Serbia, especially when voice in between telco's network went to well known
bad analog lines. I used OSLEC and I was fortunate to have Steve to complain
to, he helped patching it further after my beta testing. Not many people
wou
Hello,
I run FreeSWITCH as a PBX solution for several companies, all sharing a
single server in a "vritual pbx" deployment. Dialplans and user directories
are all separate and handled per domains. Currently, there is about 250
phones set to use it, about 200 more will be migrated soon from Asteris
softphone).
>
> So, I think that tone detection solution does not resolve my problem... Is
> there any other possibility to detect hang up without answering the call
> (using Loop Start signaling) or have we to wait until OpenZap is completely
> developed?
>
> Thanks in a
584 hangup_function() Hangup
OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]
Regards,
Ognjen
On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija wrote:
> I tried similar setup with my analog card (X100P) and I'm having same
> issue. Call is not hungup on the oz side once the caller ends. My telco
> doe
I tried similar setup with my analog card (X100P) and I'm having same issue.
Call is not hungup on the oz side once the caller ends. My telco doesn't use
polarity reversal for singalling hang up so I'm guess I'm stuck to detecting
busy tone from the telco side. I'll try to modify tones.conf accordi
Hi,
there is proto_specific_hangup_cause switch variable you can use for the cdr
i.e. You can also use SIP messages number for a continue_on_fail action
like:
Regards,
Ognjen
On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N wrote:
> Hi,
>Is there any possibilities that Freeswitch may d
Hello,
I'm using FreeSWITCH mostly as a PBX for multi tenants. Secure calling is
supported fully by FreeSWITCH and to my knowledge it is the only open-source
solution where it works w/o any hacks or tweaks.
Current major brand of phones supporting SRTP and TLS that I've tested are
Linksys and Snom
Hi,
I have multi domain, multi tenant setup configured and working.
Did you add something like
to one of the profile configs for multi-domain so FreeSWITCH can look its
configs for those domains?
Also, check if you specified domain_A for "domain_name" param in the
domain_A.xml file.
D
Diego,
good of you to listen. As for the broken core, checkout:
http://www.freeswitch.org/node/117
It says everything there is to know about disasterisk core.
Regards,
Ognjen
On Mon, Jun 30, 2008 at 6:34 PM, Diego Viola <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm sorry for my reaction... I know F
I hate HTML, XML and all the *MLs, but XML has become de-facto standard
whether we like it or not, so I use it now with FS. I don't like but I
learnt and use it.
You can't be serious that Asterisk/Digium is ok when you ask them for a
feature or anything at all? Tony offered them to rewrite obvious
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