Woof!
On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins
wrote:
>> I was wondering... does anyone make a "SIP certification" program kinda
> like a pen-tester except to find all the ways your SIP setup is broken?
> Just curious.
Here is a start:
http://interop.sipxecs.org/
It's best fo
Woof!
On Mon, 19 Oct 2009 12:07:10 -0400, Christian Löschenkohl
wrote:
> any ideas would be helpful
We run a perl script that checks if the servers are responding to
requests. It can send OPTIONS, and PING requests to various servers
periodically. If the response it gets back isn't corr
meter:
With this in place, DTMF and hangup messages traverse the nat firewall
correctly. Without it they don't. I searched the Wiki and couldn't find any
info on this parameter. Can anyone provide a description of what it does and
why it's significant that can be added to the
gh. Have I misunderstood what is
required. Is there some additional forwarding within FS required. I'm really
sorry to keep coming back but I've been wrestling with this for a long time
now and not getting anywhere.
Many thanks
Andy
_
From: freeswitch-users-boun...@
re not behind a upnp/nat-pmp router so you'll have to
manually forward everything... All the info you showed displaying the
profile status is correct.
/b
On Oct 7, 2009, at 12:44 PM, Andy wrote:
Is this because I'm using port 5060
BUT..
nat_map status, gives:
API CALL [nat_map(status)] output:
false
And there is no mention of nat detection in the startup log.
Is this because I'm using port 5060 externally?
Cheers
Andy
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-
t
helps.
Many thanks
Andy
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Woof!
On Tue, 01 Sep 2009 18:52:01 -0400, Anthony Minessale
wrote:
> there is no chance that you would not enter the conf muted the way you
> describe unless you are using an older revision of FS that had a bug in
> the parsing of the conference flags.
Perhaps some listeners are hitting the
On Wed, 19 Aug 2009 14:50:34 -0400, Matthew Fong wrote:
> I was trying to use audacity, but not sure how to tell the exact
> frequency.
Audacity can do it. Highlight the "beep" (and nothing but the beep) with the
selection cursor, then click "Analyze->Plot Spectrum..." In the "Frequency
An
Woof!
Too simple to open a JIRA with a patch (and it actually works as written):
1804: } else if (!strncasecmp(cmd, "nolinger", 6)) {
That should be an 8 as nolinger is 8 characters long.
--Woof!
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-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed voiptalk.org
2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister voiptalk.org
Andy
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
l 'sofia profile external restart' or restart the software this
fixes the problem
My questions are:
1) Why would the ping fail when the registration appears to be intact?
2) Whay would the auto re-register not work but a restart would?
This ones driving me nuts so any help
Woof!
On Fri, 10 Jul 2009 12:09:05 -0400, Anthony Minessale
wrote:
> The way it works by default is that if you send a www-authenticate, we
> *always* try to process it.
> HOWEVER, we have a accept-blind-auth sofia profile param (in fact it was
> invented just for sipX)
Then my understanding w
Woof!
It is my understanding, that if I set
in a SIP profile, it shouldn't challenge for authentication under any
circumstances.
However, if an INVITE contains a a Proxy-Authorization header from another
proxy, Sofia DOES challenge with a 407.
I'm aware one can set accept-blind-auth to
On Fri, 26 Jun 2009 00:51:34 -0400, Raymond Chandler
wrote:
> windmills
Just set them up to be right next to the output of the system's fan!
I gotta get to the patent office, quick!
--Woof!
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of concurrent calls.
Many thanks
Andy
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 18 June 2009 18:11
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Sample rate and
n/should I alter the sample rate
of the base call to 11025?
Many thanks for sorting this one for me and for all your help.
regards
Andy
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: 12 J
s for recording otherwise.
/b
On Jun 12, 2009, at 9:37 AM, Andy wrote:
Hi,
Sorry but I just can't find this in the documentation. I'm using recordFile
to record incoming messages. I'd like the audio files produced to be 11025Hz
rather than 8kHz is this possible? What setting
Hi,
Sorry but I just can't find this in the documentation. I'm using recordFile
to record incoming messages. I'd like the audio files produced to be 11025Hz
rather than 8kHz is this possible? What setting do I need to change?
M
Hi,
Every few days I'm getting this error which is causing Freeswitch to crash.
Can anyone tell me what may be causing this or how to prevent it?
2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 handle_fatality()
Caught signal 11 for unmapped thread!
Many thanks
freeswitch
install seems to crash every few days.
Also, does anyone have an example of the monit setup for freeswitch to
restart it when it fails?
Many thanks
Andy
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w use VOIPTALK.
Cheers
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of adamF
Sent: 17 May 2009 19:12
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch Drops Call aft
) Channel
sofia/external/07540526...@194.145.190.143 entering state [completed][200]
2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel
sofia/external/07540526...@194.145.190.143 entering state [ready][200]
These lines appear for calls that work and not when they don't.
Hope
phone.
It's not that digits get dropped some calls semm to handle dtmf perfectly
and others don't seem to get dtmf at all.
Can anyone shed any light opn this or suggest any solutions?
Many thanks
Andy
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Woof!
On Fri, May 8, 2009 at 4:22 PM, Lon Baker wrote:
> A truly clean install out of the gate.
We took a different approach. Rather than change what FS comes with,
we created an alternate configuration area and point FreeSWITCH to it
when we start.
Here's the script:
http://code.sipfoundry.or
server.com/myaudio.mp3) the end of the call is missing
off the resultant mp3 file.
A wild shot in the dark I know but does anyone have any experience of this
and how it might be resolved?
Many thanks
Andy
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Hi Brian,
Is NAT a known problem? Is there a work around? The messages on the lists
seem to imply other folks have this working ok behind NAT firewalls. What's
your recommendation for how I should proceed?
regards
Andy
-Original Message-
From: freeswitch-users
Hi Brian,
The freeswitch server is connect to the internet via a Cisico ASA firewall
currently running in NAT mode. I believe it's that simple but can't be sure
of the equipment between my firewall and the internet.
regards
Andy
-Original Message-
From: freeswitch-
?
regards
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 06 April 2009 14:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Calls being cut off while
s firewall related. Any
clues?
regards
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 06 April 2009 14:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freesw
ry if this is obvious but what have I done wrong?
Thanks for your help
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 31 March 2009 14:40
To: freeswitch-users@lists.freeswitc
ry if this is obvious but what have I done wrong?
Thanks for your help
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 31 March 2009 14:40
To: freeswitch-users@lists.freeswitc
Hi Brian,
1.03
Thanks
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 31 March 2009 14:27
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Calls
that intermittently,
calls keep getting cut off after a number of seconds.
I've attached a snapshot of the log at the point that the call gets cut off,
can anyone suggest why this is happening or how I can prevent it?
Many thanks
Andy
2009-03-30 11:14:43 [DEBUG] switch_ivr_play_say.c:272 switch_ivr_ph
uming
is the main issue. do you have any further tips to make this more stable and
prevent the call cut off?
Many thanks
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mathieu
Rene
Sent: 18
Woof!
Appears to be a recently fixed * bug:
0014431: Bad branch parameter value in CANCEL request
http://bugs.digium.com/view.php?id=14431
--Woof!
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x or something.
Many thanks for your help.
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 18 March 2009 14:08
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitc
get it to
reconnect with a sofia restart. I'm using the same provider and user account
as with the old version of the software. Can you suggest any reaosn why this
may be happening and how I can prevent it?
Many thanks
Andy
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Woof!
On Fri, 06 Mar 2009 01:55:43 -0500, Vikas Sharma
wrote:
> Can it be integrated with other pbx as a media server?
Yep, it sure can:
http://sipx-wiki.calivia.com/index.php/SipXivr
--Woof!
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Woof!
On Mon, 02 Mar 2009 19:32:53 -0500, Steve Underwood wrote:
I just had a look through that patent. Its amazing. There is a lot of
> focussed descriptive text, but a patent only really consists of its
> claims. Those claims are astonishingly open-ended, and characterise what
> people had be
Woof!
On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote:
> NO. You want something that people THINK exists and works well...
> Reliable human/voice detection doesn't exist in ANY form.
I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way
to do it. It works rathe
Woof!
On Thu, 12 Feb 2009 17:20:18 -0500, Anthony Minessale
wrote:
> So I wonder what about the distro you are using that makes the same exact
> code not work?
> maybe the GCC ?
Possibly. A recent (last year?) GCC change caused some order of operations to
change, and so code that inadverten
8, 2009, at 8:37 AM, Andy Ayers wrote:
Hi,
Is the bitrate, sample rate or format of the audio stream created by
session.recordFile configurable at all? Apologies if I've missed something
in the docs.
cheers
Andy
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Hi,
Is the bitrate, sample rate or format of the audio stream created by
session.recordFile configurable at all? Apologies if I've missed something
in the docs.
cheers
Andy
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Woof!
On Wed, 07 Jan 2009 18:41:31 -0500, Michael Jerris wrote:
> you can however use mod_limit to implement this yourself with dialplan
> logic as long as it is used before all calls to the conference (it
> wouldn't work for outbound calls from the conference without a little
> bit of thought)
Woof!
On Wed, 07 Jan 2009 14:31:54 -0500, Anthony Minessale
wrote:
> we don't currently have anything like that.
Okay. Just wanted to make sure I wasn't missing something obvious.
--Woof
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Woof!
Does FS have a way of limiting the total number of conference legs on a
box? I am aware that each individual conference profile can have a
"max-members" param, but what I'm looking for would span multiple
conferences, with a maximum leg limit per server, regardless of the per
confer
Woof!
On Mon, 05 Jan 2009 23:43:26 -0500, jonathan augenstine
wrote:
> According to RFC 3515 there are no BYE messages in the protocol exchange.
Once the REFER is completed (as determined by a final response returned in
the NOTIFY SIPFRAG from the REFER), the original dialog can be torn dow
ss the firewall/router? The router is currently set
to allow all traffic.
Many thanks for any help you can give.
regards
Andy
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Woof!
On Mon, 22 Dec 2008 16:46:14 -0500, Brian West wrote:
> When we convert them from 48k we can lower the vol a bit more we are
> already doing it slightly.
>
The prompts we are using aren't from the FS set. It's not a matter of
adjusting the prompts, they've work fine for G.711 for years
Woof!
I've noticed that the percieved volume of prompts recorded at l...@8000 is much
louder (to the point of distortion) when played back via G.722 on Polycom
phones, vs when played back via G.711. The same prompts are also slightly
louder when played back on SNOM phones via G.722 vs G.711, b
Woof!
On Mon, 15 Dec 2008 13:16:32 -0500, Raymond Chandler
wrote:
> if freeswitch.history isn't a log, what is it? seems to me taht it's a
> log of what commands you've run recently... it's definitely NOT a
> database
Actually, I the readline/history library uses it to determine the comman
Woof!
On Mon, 15 Dec 2008 11:56:53 -0500, Brian West wrote:
> I can't figure out why the log file would need to be in the db folder...
I think you misunderstand.
It's these files:
freeswitch.history
freeswitch.pid
freeswitch.xml.fsxml
That I feel would be better off in the db folder.
Woof!
On Sat, 13 Dec 2008 18:52:59 -0500, Michael Collins wrote:
> All you'd have to do is modify the logfile.conf.xml file and pick a new path
> for your freeswitch.log file...
I agree. I had discovered this option and considered it as a workaround. Then
I also found that mod_xml_rpc was a
Woof!
It appears that FreeSWITCH writes
freeswitch.history
freeswitch.log
freeswitch.pid
freeswitch.xml.fsxml
to the -log directory.
Is there a way to put the files other than freeswitch.log into the -db
directory instead?
In my environment we archive and rotate everything in t
VOIP Providers?
Please have a look at the wiki
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
In which country do you need a provider?
So far I got every provider I tried (5) to work with freeswitch; even with
FS behind NAT.
Andy Ayers schrieb:
> Hi,
>
> Can anyone recommend
Woof!
Anthony wrote:
> I asked because I was wondering if you could test the scenario I described to
> compare what happens to a call from x-lite being deflected since I know for a
> fact it was working.
Woof wrote:
> FS still doesn't hangup on the original call after the REFER is completed.
I
Woof!
On Fri, 21 Nov 2008 14:49:35 -0500, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> can you try uuid_deflect again on latest,
Done.
> I see we give up to easy, "on 180 instead of final response in sipfrag"
> the one i tested didnt send 180 so i forgot about the possibility.
>
> post a full
Woof!
On Thu, 20 Nov 2008 19:06:09 -0500, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> When i was testing I called into park ext with fs and did
> show channels
> uuid_deflect
> sip:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
>
> and when it sends me the notify the command returns and prints the sipf
Woof!
On Thu, 20 Nov 2008 19:06:09 -0500, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> I wonder if sofia is not matching the notify to the dialog so we are not
> associating it with the channel.
> I know for a fact sofia tears it down for us when it gets the notify.
>
> do you have x-lite/eyebe
Woof!
On Thu, 20 Nov 2008 15:14:52 -0500, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> how were you calling the deflect the way that had no change?
> every time i tried it sofia has taken down the channel once it completed.
Hmm...
dialplan:
Call from "207" to "[EMAIL P
Woof!
Thanks for the changes!
On Wed, 19 Nov 2008 21:09:19 -0500, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> try latest code and see how that works.
FreeSWITCH Version 1.0.trunk (10481)
No difference with just "deflect"--the call does not clear when the REFER is
completed, nor are there a
Woof!
When using "deflect" on an inbound answered call, I notice that the FS channel
stays connected as long as the original call exists, and does not send a BYE to
the original call even after receiving a NOTFIY with a final response fragment.
In addition, while FS gets those NOTIFY's, I haven
iated.
regards
Andy
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Woof!
On Mon, 03 Nov 2008 08:55:11 -0500, Klaus Teller <[EMAIL PROTECTED]>
wrote:
> Hi Folks,
>
> Just to let you know that we are working on a library for connecting to
> the Freeswitch via the socket interface. We plan to release it under
> LGPL as soon as it's somewhat robust.
You may
Woof!
Is there any updated projection of when 1.0.2 will be released? Last I
can find mention of it is over a month ago (http://freeswitch.org/node/143)
Thanks,
--Woof!
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Woof!
On Wed, 29 Oct 2008 14:31:36 -0400, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> update and try again ;)
Anthony, you are just too fast! Mike was saying one thing, and you
checked in the change it while I was replying to him!
Updating now.
--Woof!
Woof!
On Wed, 29 Oct 2008 14:18:53 -0400, Michael Jerris <[EMAIL PROTECTED]> wrote:
> They should already be on the initial events. Take a look at the raw
> output, you probably were taking them out of a later event.
Nope. Initial event. No variable_* are reported. Using netcat:
Connection
Woof!
On Wed, 29 Oct 2008 09:10:32 -0400, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> by default only some hand-picked events have all the variables due to
> people complaining that they had too much info ;)
Sending the same info over and over does seem counter productive.
> if you want
Woof!
I used to get lots of variable_* lines when using socket_outbound. They
have disappeared. Is there something I need to configure to get them back?
--Woof!
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Woof!
On Mon, 13 Oct 2008 11:02:49 -0400, Kristian Kielhofner
<[EMAIL PROTECTED]> wrote:
> Instead of modifying the URI, why not attach your own header:
> X-conf-pin: 1234
> I haven't done it yet, but it might be as simple as:
> action application="conference" data="confname+${sip_X-conf-pin}
ppreciated.
Many thanks
Andy
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Woof!
On Thu, 11 Sep 2008 15:57:26 -0400, Michael Jerris <[EMAIL PROTECTED]> wrote:
> Fixed in svn r9527, thanks for the report.
Verified. Thanks.
--Woof!
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Woof!
We've been seeing core dumps on FreeSWITCH shutdown. I just updated to
today's revision (r9526) and still see them:
freeswitch: src/switch_core_hash.c:59: switch_core_hash_destroy: Assertion
`hash != ((void *)0) && *hash != ((void *)0)' failed.
Core was generated by `/usr/local/freesw
Woof!
On Mon, 18 Aug 2008 14:25:45 -0400, Bruce McAlister <[EMAIL PROTECTED]> wrote:
> Does anyone have any idea's on the error listed below?
> /opt/SUNWspro/bin/cc -w -DMULTIPLICITY -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64 -D_TS_ERRNO
> -I/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE -DEMBED
Wof!
On Fri, 25 Jul 2008 13:09:45 -0400, Boris Krivonog <[EMAIL PROTECTED]> wrote:
> Hi all!
>
> I'm using event sockets to remotely drive freeswitch. Is there a read like
> functionality for speak that can be used with event sockets, for example:
>
One way to handle DTMF collection with event
Woof!
On Wed, 09 Jul 2008 15:07:56 -0400, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> silence_stream://[,]
>
> eg
> silence_stream://1
>
> will generate 10 sec of absolute zeros
Nice! Pefect for my needs. Thanks.
--Woof!
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Woof!
On Wed, 09 Jul 2008 12:49:51 -0400, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> you can try 1,1 instead of 0,0 which is still silent
Actually, it isn't. It is quite audible on several phones I've played with
(most likely due to the uLaw companding adding a "step" to each voltage, and
Woof!
On Wed, 09 Jul 2008 11:24:20 -0400, Michael Jerris <[EMAIL PROTECTED]> wrote:
> What do you get now when you try to do this?
When I do "tone_stream://%(1, 0, 0)", expecting 10 seconds of silence,
it instantly completes the command without error. It doesn't take 10 seconds
;-)
--Woo
Woof!
On Tue, 08 Jul 2008 19:13:24 -0400, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> try now =D
Ahh, now there is joy!
Can I trouble you to also make "tone_stream://%(150, 0, 0)" (or something
similar) also work? I can forsee a need for injecting short pauses into a
sequence of prompts,
Woof!
On Tue, 08 Jul 2008 15:27:02 -0400, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> update and try again.
Alas, no joy. Break still won't wake it up.
svn version 8933
--Woof!
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Woof!
On Tue, 08 Jul 2008 13:27:40 -0400, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> Try latest trunk, I added support for what you want.
Thanks. I just tried it. DTMF event reporting now works...but I cannot
"break" out of the sleep command once it is started.
--Woof!
_
Woof!
Using mod_socket (outbound mode) on svn version 8911, after the call is
answered, I'm having difficulty with generating silence periods during which I
want to be able to detect DTMF events.
If I use:
sendmsg
call-command: execute
execute-app-name: sleep
execute-app-arg: 5000
It won't de
Woof!
On Thu, 05 Jun 2008 10:17:14 -0400, Ken Rice <[EMAIL PROTECTED]> wrote:
> The whole problem is the
> Patents held by The G729 Consortium, and 2 other companies...
>
Has anyone ever gotten a list of the actual patent numbers in question? I've
tried on several occasions and have always bee
Woof!
Using an event socket on an answered parked call, I am trying to use "deflect"
to transfer the call off of FreeSwitch to a SIP destination, like this:
sendmsg
call-command: execute
execute-app-name: deflect
execute-app-arg: sip:[EMAIL PROTECTED]
Alas, our proxy will challenge the REFER t
Woof!
On Fri, 23 May 2008 20:02:46 -0400, Brian West <[EMAIL PROTECTED]>
wrote:
> Please update to at least rev 8564 and try again.
Will do next Tuesday. Thanks for the quick fixes. Enjoy the long weekend
(for those of you who are US based, at least!)
--Woof!
Woof!
On Fri, 23 May 2008 17:57:59 -0400, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> see if just async and not async full works
Just tried that. No, it doesn't work either.
--Woof!
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Woof!
On Fri, 23 May 2008 17:13:59 -0400, Brian West <[EMAIL PROTECTED]> wrote:
> How are you calling the socket application for the outbound event
> socket connection? If you put it in async you should get those events.
I've tried it various ways, but this is the one I'm currently using:
Woof!
I'm experimenting with event sockets (inbound, if it matters), and was playing
dialtone via the gentones app, like this:
sendmsg
call-command: execute
execute-app-name: gentones
execute-app-arg: %(1, 0, 350, 440)
\n\n
When I pressed digits on the phone, instead of the DTMF events show
Woof!
On Wed, 09 Apr 2008 15:12:33 -0400, Michael Collins <[EMAIL PROTECTED]> wrote:
> to the best of your knowledge do the
> Nortel's support TBCT?
Yes, they do. But only on the National ISDN 2 flavor of PRI (as of 10 years
ago,
anyway!). There is also a settable "choke" limit on how many ca
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