side nothing was changed which used to work fine
with FS
Chris
On Thu, Jun 25, 2009 at 4:15 AM, Brian West
wrote:
Please open a jira and attach sip traces of register and phone calls.
/b
On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote:
I’ve been using FS as a gateway to our OCS server for
Are you the owner of Versafon?
/b
On Jun 25, 2009, at 10:24 AM, paul.degt wrote:
> You can use FS XML Curl - FS sends XML CDRs to a web server of your
> choice, and there you do whatever you want with these CDRs, like store
> in a database.
> There are also pre-built solutions available, check
Fixed revision 13948.
/b
On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote:
Done, added as issue SFSIP-157.
Regards,
Peter Olsson
Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] För Brian West
Skickat: den 25 juni 2009 10:16
Till
work fine
with FS
Chris
On Thu, Jun 25, 2009 at 4:15 AM, Brian West
wrote:
Please open a jira and attach sip traces of register and phone calls.
/b
On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote:
I’ve been using FS as a gateway to our OCS server for some time.
It’s used just for
check that s is nil.
/b
On Jun 24, 2009, at 8:12 PM, John Wehle wrote:
What's the recommended way to check if the session constructor was
successful (i.e. the number could be dialed)?
Brian West
br...@freeswitch.org
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Chances are you need to get var us-ring then use that to set the
transfer_ringback
/b
On Jun 24, 2009, at 9:20 AM, Diego Toro wrote:
Hi Brian, with Session.SetVariable("transfer_ringback", ${us-ring});
I have message: "[CRIT] switch_channel.c:633 Invalid data ($
{tr
al plan the call is stablished.
I have FS rev 13750 running on Windows.
This is a issue or I don't use properly transfer_ringback variable ?
Diego
Brian West
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which have this feature?
Lars
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ing [en]
2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-goodbye.wav] (en:en)
I assume the vm-record_message.PCMU is the file that will be created
to record the voicemail. Is that correct and how can I fix this?
Thanks!
Brian West
br...@freeswitch.or
all on the event API
2 – I used sleep 18 (3 mins) see rule below.
3 – failed - because the rule is executing a sleep command and I
cannot break in with my redirect.
Brian West
br...@freeswitch.org
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Jun 23, 2009, at 7:51 AM, Mark Campbell-Smith wrote:
Because of this, I never get audio. Any ideas how to fix this?
Thanks!
Brian West
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't hijack
threads please.
thank you in advanced,
best regard,
mashudi
Brian West
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what is 8000? is it local or is it a remote endpoint?
/b
On Jun 22, 2009, at 3:01 PM, Tim B wrote:
originate sofia/default/8...@192.168.10.35 &txfax(storage/fax/
test.tif)
Brian West
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spec now to try
that.
Thanks guys!
jlc
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what is 242424? If its a locally registered user you should be using
a % instead of an @
/b
On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote:
Hmm thamks. I tried it and it doesn't work out of the box. Here are
my logs: http://pastebin.freeswitch.org/9454
Thanks,
Max.
Brian We
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Brian West
br...@freeswitch
ly it ends with:
Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/
freeswitch-1.0.4-1-root-rpmbuilder
RPM build errors:
Which doesn't help:)
Thanks!
jlc
Brian West
br...@freeswitch.org
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this?
I dialing a mobile number on this sometimes it works... Sometimes it
destroys the call [CALL_DESTROY]
Thanks
Brian West
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bingo! :P
/b
On Jun 22, 2009, at 4:06 AM, Darren Schreiber wrote:
Ignore this thread. Apparently I was stripping sip: from the prefix.
I guess you have to specify sip: before utilizing fs_nat and fs_path
variables.
My bad.
Brian West
br...@freeswitch.org
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h");
session.execute("say","th number pronounced 1346523");
session.execute("say","th number pronounced 21");
session.execute("say","th number pronounced 11");
session.execute("say","th number pronounced 101");
it was actually tony but who cares! ;)
/b
On Jun 22, 2009, at 12:08 AM, Mathieu Rene wrote:
13887 created by brian on 21 June 2009, 21:52:31 -0500 (75 minutes
ago) (patch) move this to debug and profile->debug so that its not
on unless you enable the profile debug also.
That should
nobody authenticates on the request URI... you're focusing on the
wrong thing... you'll need from-domain and/or from-user I suspect.
/b
On Jun 21, 2009, at 7:16 AM, Jan Kubr wrote:
> Creating a separate sofia profile just for this gateway definitely
> works, just wondering whether there is a
They usually will not auth on the RURI... I recommend you set the from-
domain on your gateway... I think thats really what you need.
/b
On Jun 21, 2009, at 6:39 AM, Jan Kubr wrote:
I have found this: http://jira.freeswitch.org/browse/MODENDP-184
Thanks to which I know that adding
to the
http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
/b
On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote:
> Hi!
>
> I am trying to email from
> 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore
> original codec.
> 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:
Really it shouldn't have changed unless you wiped your configs. The
reason it can't work with auto-nat is you're not behind a natpmp or
upnp router thus you're going to have to set them manually use stun.
You can not use stun for rtp-ip or sip-ip, just for ext-sip-ip and ext-
rtp-ip once t
There actually are issues between the GPL and MPL :P
/b
On Jun 20, 2009, at 11:12 AM, paul.degt wrote:
> source software there should be absolutely no difference between GPL
> and
> MPL.
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I still say why not MPL or at the very least MPL/GPL?
/b
On Jun 20, 2009, at 9:37 AM, paul.degt wrote:
> Yes, that's one of the reasons. Another point is that GPL v.3 is
> defined
> more clearly from legal perspective, at least from our legal adviser
> point of view.
>
> Diego Viola wrote:
>>
Depends on what you're doing ... or not doing...
/b
On Jun 19, 2009, at 3:20 PM, Matthew Fong wrote:
> With yesterday's trunk and also a release from 2 weeks ago, I
> noticed that my freeswitch process as it ran was eating up more and
> more memory. At the end of the day it was using 75% of
No right now you can not legally transcode G729 in FreeSWITCH, PERIOD!
/b
On Jun 19, 2009, at 2:11 PM, JuanMa wrote:
> Yes, it can do transcoding. Transcoding isn't the problem to my
> architecture, my problem is the codec negotiation between FS and
> Endpoints.
>
_
If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT
issue.
/b
On Jun 19, 2009, at 1:38 AM, Matthew Fong wrote:
the script is not part of a session or dial plan. :(
On Thu, Jun 18, 2009 at 11:31 PM, Jason White
wrote:
Mathieu Rene wrote:
> data="failure_causes=user_bus
I would like to thank everyone for Dinner... we had a great time...
now MORE CODE!!!
/b
On Jun 18, 2009, at 7:51 PM, Michael S Collins wrote:
> FYI, the devs report that they are at the restaurant! Last chance to
> pitch in and feed the troops. :) hit the paypal button on the main
> FreeSWITCH
The call rates we support are 8, 16,32 and 48k
/b
On Jun 18, 2009, at 1:01 PM, Andy wrote:
Thanks Brian,
So, just to calrify will the base call always be 8kHz?
On a related note, do you happen to know the bitrate of each open
channel/live call? Is it 16 kilobits per second like the
look in mod_shout you'll see my calculations.. I think it has to be
multiples of 16 if I recall.
/b
On Jun 18, 2009, at 1:01 PM, Andy wrote:
On a related note, do you happen to know the bitrate of each open
channel/live call? Is it 16 kilobits per second like the recorded
audio? I need
On Jun 18, 2009, at 11:54 AM, Andy wrote:
1) I notice that when I change the sample rate it automatically
changes the bit rate too. I understand why this is the case but
wondered if it was just as easy to be able to control the bitrate as
well as the sample rate.
If you're talking about
Please post bugs to http://jira.freeswitch.org
/b
On Jun 18, 2009, at 11:20 AM, Victor Toofic wrote:
> Hi all!
>
> I'm having some troubles with call quality using conferences. The
> scenario is like this:
>
> An agent makes a call to freeswitch and enters in a conference room
> waiting for outb
ch in on that also please let me know. I'm paying this out
of my pocket.
Thanks,
Brian
On Jun 17, 2009, at 1:56 PM, EdPimentl wrote:
I will match the 150.00
Best regards,
-E
CEO and Founder
Gpro.ws
http://Twitter.com/edpimentl
http://TwebEX.com (Twitter Based Online Web Conferenc
I suspect you're going for TALK and NOTALK as the event names?
its CUSTOM conference:: maintenance
/b
On Jun 18, 2009, at 8:00 AM, Steven Brown wrote:
Hi,
I have been trying to pick up TALK and NOTALK events but with no
success, I have enabled VAD for "both" in my config and the rtp is
st
You're trying way too hard. CALL Rejected gives us exactly ZERO to go
on... We are all trying really hard to help you but at some point we
just can't help anymore. Please make sure you post debug logs to
pastebin and join IRC. This email back and forth over something like
this just takes
I no longer need your configs... I didn't try to put
stun:stun.freeswitch.org in sip-ip or rtp-ip because I know you
shouldn't. We clearly can not try to do a stun request in either of
these fields because you can't bind to IP's that aren't directly on
the machine... so do as per the confi
Ok you both didn't notice you CAN NOT put stun:stun.freeswitch.org in
rtp-ip, thats the problem. It clearly says IP ADDRESSES ONLY in the
comments. DO not use $${external_rtp_ip} for rtp-ip either :P
/b
On Jun 17, 2009, at 3:10 PM, Raul Fragoso wrote:
I can confirm the same issue, but it
I need in one of your boxs... there is no way this is doing this
unless you are putting stun:stun.freeswitch.org into the ext-rtp-ip or
sip ip... which could make it trigger the ipv6 check since its just
looking for : in the ip address. And stun: has that.. so you're
triggering it... tar u
And a
generic ETA of those deliverables?
Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/
On 17-Jun-09, at 23:35, Brian West wrote:
Guys I'm tossing in $250 dollars of my own money on this ... who is
goi
Thanks for the match.
/b
On Jun 17, 2009, at 1:56 PM, EdPimentl wrote:
I will match the 150.00
Best regards,
-E
CEO and Founder
Gpro.ws
http://Twitter.com/edpimentl
http://TwebEX.com (Twitter Based Online Web Conference Platform)
http://TwitrShare.com (Send Picture and Message to Tweet C
mod_pocketsphinx will be there still as will mod_flite.. this lets you
offload ASR and TTS to another server in a standard way. ScribleJ
hasn't really helped me with mod_pocketsphinx. The out of grammar
segfault is now gone if you update ;)
/b
On Jun 17, 2009, at 1:50 PM, Mitul Limbani w
Its not possible right now but you could if you enable the config
option and apply the tag... its something I have thought about adding
but wasn't high on my list.
NTATAG_SIPFLAGS(MSG_FLG_COMPACT)
http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6
t to Arsen.
Thanks,
Brian
Begin forwarded message:
- Forwarded Message
From: Arsen Chaloyan
To: UniMRCP
Sent: Wednesday, June 17, 2009 10:57:30 PM
Subject: [UniMRCP] Open source ASR and TTS plugins
Anybody interested in the development of open source ASR and TTS
plugins for UniMRCP se
Thats one way to put it ;)
/b
On Jun 17, 2009, at 12:26 PM, Diego Viola wrote:
> Applications are the ones in mod_dptools and FSAPI are mod_commands
> API right?
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Its clearly telling you that context features doesn't exist... did you
remove the context tags around your extension so that it would be in
the correct context? Review the default config again.
/b
On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote:
> Context features not found
The link would be helpful.
/b
On Jun 17, 2009, at 1:14 AM, selva kumar wrote:
Hi Michael,
I have pasted the freeswitch logs as requested in
(pastebin.freeswitch.org)
Thanks
Sam
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No, we haven't done the solution file for the module on windows but
the lib has been done on windows. It still need more testing and such
but its functional.
/b
On Jun 17, 2009, at 8:17 AM, Peter Olsson wrote:
I can see that the UniMRCP libs have been added to FS lately, I was
just wonde
On Jun 17, 2009, at 8:14 AM, Jim Burke wrote:
> IMHO, as FS is a B2BUA the new leg should state ownership in the
> SDP. Add to this the fact the IPV6 is blindly copied from leg 1 and
> the IP address was not decoded correctly there does appear to be a
> defficiency in the code.
I don't th
While the header looks valid it should be an X-Header then it would
show up.
/b
On Jun 17, 2009, at 2:32 AM, Ken Rice wrote:
> I doubt that header is exposed since its not a standard sip header.
> However
> you could probably patch mod_sofia to expose it without too much
> trouble...
> How
Right which is what you have to do... I haven't been able to reproduce
the issue... which is odd.
/b
On Jun 17, 2009, at 2:36 AM, Jason White wrote:
> Jason White wrote:
>
>> The symptom is the following line in outgoing SIP messages while
>> attempting to
>> establish a call to a gateway v
can you update and try that again?
/b
On Jun 17, 2009, at 12:00 AM, paul.d...@gmail.com wrote:
> 13564
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Its not an error its a warning and you don't have your ACL's
configured correctly. You're trying too hard! :) set auth-
calls=false on the profile.
/b
On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote:
>
> Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on
> FS B
COPY paste fail :)
something like that as per the example.
/b
On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote:
>
> How can sofia profile can call ACL?
> Can you give me an example?
> Like this?
>
> I put this on external profile
>
> "/>
> "/>
>
Now you have to tell the sofia profile to use that ACL
/b
On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote:
How can i turn off authentication? This is my acl.conf.xml on
192.168.0.105
On 192.168.0.4
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Shouldn't have really changed any behavior at all... What svn rev are
you on?
/b
On Jun 16, 2009, at 5:50 PM, paul.degt wrote:
> API CALL [global_getvar()] output:
> external_ssl_enable=false
> external_tls_port=5081
> external_sip_port=5080
> external_auth_calls=false
> internal_ssl_dir=/var/
please see MODOPAL-10 on jira.
/b
On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote:
Hello, all. I'm currently playing around with a new install of
Freeswitch and wanted to try out mod_opal. Below are the current
SVN builds for opal, ptlib, and freeswitch. I end up with the
followin
This should be a huge clue... what might be your providers name?
Seems something is missing here or you have the settings wrong.
/b
On Jun 16, 2009, at 9:58 PM, Ing. Edwin Villarreal wrote:
DNS Error [503].
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Turn off authentication or use ACL's
/b
On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote:
> Is there another way to manage the gateway with the caller id of the
> user
> not the gateway user id and is there a gateway that doesn't need a
> username
> and password?
_
I need sip traces... also can you guys register to my dev box?
dev.bkw.org with default user/pass try 1009 thru 1015 please.
/b
On Jun 16, 2009, at 8:17 AM, Seven Du wrote:
What's wrong of the contact string? 639(snom) works but 637(zoiper)
doesn't.
"user" >
"seven" >
Ok i'll have to se what I can do about reproducing this issue now that
I have more info on how its happening.
/b
On Jun 16, 2009, at 7:40 AM, dujinfang wrote:
Almost caught you on IRC Mike.
Our server is in a NAT'd network and all agents registered in the
same LAN. I can remotely register
Why not just keep the agent off hook.. in park state... then just
playback ringing before you bridge?
/b
On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote:
> Hello Michael,
>
> I want the phone be ringing, just for acoustical feedback reasons.
>
> But what if I
>
>* transfer it to the same us
Can you please put it back to auto-nat and email me the output of
global_getvar from the CLI so I can see what it detected?
/b
On Jun 16, 2009, at 7:18 AM, paul.d...@gmail.com wrote:
> Solved by replacing "auto-nat" with public ip in public profile
> "external_sip-ip" and "extrenal-rtp-ip" par
Pretty useless without 64bit support.
/b
On Jun 15, 2009, at 9:58 PM, EdPimentl wrote:
> As of April 09 it did not support 64bit not sure if it has been
> added since then.
> -E
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click on the AA button? :)
/b
On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:
>
> What is the best way to have this done? Move the call to park and then
> retransfer again with intercom, or is there a better solution?
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I don't think V8 will work on 64bit yet will it?
/b
On Jun 15, 2009, at 5:45 PM, EdPimentl wrote:
> Actually these new SSJS engines such GoogleV8 and other such as JAXER
> Bring a entire new way of building robust webapp/desktop app/ mobile
> app like it has never been built before...
>
> For
On Jun 15, 2009, at 5:08 PM, Stephen Crosby wrote:
> I'm actually much more interested in the HTTP library and a few
> other components than MySQL. Freeswitch's spidermonkey CURL library
> doesn't provide returned HTTP status codes and JSEXT does.\\\
Patch it! ;)
>
>
> That said, I'm still
Survey says ... "execute the ring_ready application"
/b
On Jun 15, 2009, at 2:50 PM, Ron McLeod wrote:
Something to consider is how long will be PSTN allow the call to
remain un-answered.
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.o
To: <"user"
Can you reproduce this or let us in your box to look at it... someone
else reported this but I have yet to be able to reproduce it.
/b
On Jun 15, 2009, at 2:41 AM, seven wrote:
Hi,
I'm on version 13524, call from zoiper is ok, but when call zoiper,
i
For clarification ... Read section 3.2 and 3.3 of the MPL 1.1
The simplest way I can describe it is how it was described to me
"What's yours is yours and what's mine is mine!".
/b
On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote:
> I have some confusion about FreeSWITCH's Mozilla Public L
Then you're free to write your own directory hook and plug it directly
into the database how ever you wish. Look how XML_CURL uses this
interface.
/b
On Jun 13, 2009, at 5:28 PM, Adam Wilt wrote:
> Thanks. I would really like mod_voicemail to be database driven,
> instead of by XML and cU
...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Brian West
Sent: Saturday, June 13, 2009 1:08 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in Dialplan documentation?
Yes look at the default dialplan... you should note that its
Yes look at the default dialplan... you should note that its in the
default only.
/b
On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote:
At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall,
near the top under From Dialplan, it says:
Bridge the incoming call to extension 100 and 101.
the way I'm
trying to use it, and the best way to approach this?
Thanks,
Adam
Brian West
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so life is good. :-)
-- John
Brian West
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U
tp://wiki.freeswitch.org/wiki/RTP_Issues
Brian West
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on the wiki
or Google. I did try before asking this list. My query to Google was
“Freeswitch unregister”. That was the best I could do given my
limited knowledge.
Thank you for the help. I’ll learn eventually.
Lars
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http
#x27;d like the audio files
produced to be 11025Hz rather than 8kHz is this possible? What
setting do I need to change?
Many thanks
Andy
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Brian West
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Freeswitch-use
e my work, once completed, in FreeSWITCH, can
you provide me the guidelines and / or eligibility criteria to do
so, any link on FS site etc.?
You post your work to our issue tracker http://jira.freeswitch.org
Thank you.
Brian West
br...@freeswitch.org
-- Meet us at ClueC
red REFER too but there seems to be even less support for
that.
ACK really? thats sad!
If I can't get the socket-sharing upgrade working then I will fall
back to
this - and peers which don't support the 302 response (or more likely,
don't authorise it) will just get
See I knew that was a bit of crack :P, Good to hear its working like
it SHOULD now!
/b
On Jun 11, 2009, at 9:21 PM, Lars Zeb wrote:
snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX-
but delivers XX to FS.
Thanks Brian
Brian West
br...@freeswitch.org
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You should be running 7.1.35 or higher.
/b
On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote:
snom320-SIP 6.5.17.
Brian West
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What firmware?
/b
On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote:
It’s a SNOM 320.
Brian West
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http
Only if they have an as xml modifier
/b
On Jun 11, 2009, at 6:25 PM, João Mesquita wrote:
Nik, I am a noobie and all, but most API responses can come as xml
just by adding "as xml" at the end of the call.
jmesquita
Brian West
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phone numbers numeric only.
Brian West
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You could also attach to it with GDB and see if its hanging somewhere
else.
/b
On Jun 11, 2009, at 4:20 PM, Michael Collins wrote:
Do they show on "show calls"? Or do they show up on "show channels"
only? Just curious to see if they were bridged or not.
ation id can then be used to park the call in
the proper fifo.
Brian West
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what?
On Jun 11, 2009, at 4:16 PM, NOx-WHV wrote:
Can you just announce b2bua.
Brian West
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http
x27;t. Its not possible because we are a b2bua and you have
already negotiated the keys between the endpoints and FreeSWITCH and
when you redirect the media neither phone can decrypt the packets
correctly.
I use a SNOM hardphone and a phonerlite softphone.
Thanks for sour help!
Don't do it! Doing that stuff is highly silly.
/b
On Jun 11, 2009, at 2:49 PM, Lars Zeb wrote:
There’s got to be a better way. Any suggestions?
Brian West
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Frees
="^100[09]$" break="on-true">
and
expression="^100[09]$" break="on-true">
and
expression="^100[09]$" break="on-true">
Brian West
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make sure you set it before the bridge.
/b
On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote:
Bridge
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Brian West
Sent: Thursday, June 11, 2009 7:24 AM
To: freeswitch-users
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Brian West
br
respond will do exactly that... try just hangup
/b
On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote:
Hi Team,
I'm still in need of a way to reject a call without answering it. I
very much appreciate your help.
Klaus.
Brian West
br...@freeswitch.org
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