try these drivers:
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.7.smg_pri.4.tgz
Mike
On Dec 29, 2009, at 6:17 PM, Jerry Richards wrote:
> I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the
> bug is still present. Would libpri possibly help? I'm currently u
This means there was no sdp sent. Did you confirm this with siptrace?
On Dec 29, 2009, at 10:37 AM, Lei Tang wrote:
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong,
following code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
There is no need for you to show us traces. The fact that you are
using proxy media is enough to know that the issue is with your
device. If you look at the full sip trace you will see the same.
Mike
On Dec 29, 2009, at 10:10 AM, Lei Tang wrote:
The phone I'm using is x-lite and wxCommu
the build system already has targets for all of this and there are tarballs you
can manually download and extract as well that are located in
http://files.freeswitch.org/. if you NEED packages, you will have to wait
until that work is complete or figure out what the error is.
Mike
On Dec 28,
The issues you ran into are probably sorted out now. Give it a try and if its
still not working, post the build errors.
Mike
On Dec 28, 2009, at 8:15 AM, Dome Charoenyost wrote:
> Oh...sory i forgot chismas and new year.
> if someone come to thailand please let's me know :)
>
> BG
> Dome
There is no such thing as freeswitch 1.5. Have you tried latest svn
trunk to see if this behavior is the same?
Mike
On Dec 23, 2009, at 7:49 AM, Lei Tang wrote:
Hi all, I'm using FS 1.5, doesn't somebody known something about
this problem?
My scenario is :
A(FreeSwitch)
Of course there is a way. Depending on the interface your looking at
either a freeswitch endpoiny module or an openzap module.
Mike
On Dec 23, 2009, at 4:54 AM, Kristoff Bonne
wrote:
Hi Rupa,
None. That's exactly the point.
Everything has to be done over the usb "HID" interface.
I've
Sounds right to me, just assign it to me if it lets you
Mike
On Dec 23, 2009, at 12:03 AM, "Joseph L. Casale" wrote:
>> For the path in the dialplan I don't think we have any right now but
>> file a bug on jira and I can try to add them. As for something in
>> the
>> script itself that is a
That being said, ulaw l16 alaw will cause degredation and any other
modifications such as volume adjustment in this path will make it
worse. Tha being said that does not sound like what you are
experiencing
Mike
On Dec 22, 2009, at 10:29 PM, David Knell wrote:
> On the other hand, a u-la
We expect the g729 sometime very soon, weeks not months away.
Mike
On Dec 22, 2009, at 7:45 PM, Rupa Schomaker wrote:
> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji
> wrote:
>> Hello people,
>>
>> Can someone please clear the following ambiguities with codecs:
>>
>> Are we definitively able t
If your seeing the trafic in ngrep bit not in sip trace in Sofia when
enabled, your firewall is blocking the traffic
Mike
On Dec 22, 2009, at 5:20 PM, Michael Collins wrote:
On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote:
Yes, the internal profile exists.
Name
Not sure if we have an option to disable info. Even without this,
dtmf should go across the bridge fine. Please open up a bug on jira
about this
Mike
On Dec 22, 2009, at 6:40 AM, Peter P GMX wrote:
> Hello,
>
> in a bigger installation with some thousand endpoints in the field we
> see, t
For the path in the dialplan I don't think we have any right now but
file a bug on jira and I can try to add them. As for something in the
script itself that is a bit more work but if anyone has a patch to
inject some vars into scripts like that it would be a nice addition.
Mike
On Dec 21,
If this is using prid it also requires the latest drivers from
sangoma. I am pretty sure these are just in dev snapshots not release
drivers yet. Something 3.5.8.6 or later iirc.
Mike
On Dec 21, 2009, at 7:52 PM, Brian West wrote:
You know that warning is meaningless. Search the archive
This is a total work in progress that has not even merged into tree. So it is
not "known" to work or not work anywhere. Patches to correct issues are
welcome.
Mike
On Dec 21, 2009, at 3:49 PM, Joseph L. Casale wrote:
> >Working on it, moving the repos around to do this right...
> >
> >http:/
Working on it, moving the repos around to do this right...
http://jira.freeswitch.org/browse/FSBUILD-218
Mike
On Dec 21, 2009, at 2:56 PM, Joseph L. Casale wrote:
> So the spec from trunk says “Soundfiles are moving into a separate spec”
> but I can’t find this spec anywhere in svn?
>
> Anyone
The best help to track this down is to try to identify the specific
svn revision that caused the issue and to supply a full freeswitch
debug with sip trace.
Mike
On Dec 19, 2009, at 3:31 AM, Jason White wrote:
> Revision 15904 is fine, but after upgrading to revision 16003 I get
> the
> f
t;
> Thank you,
> Dorn B.
> From: Michael Jerris
> To: freeswitch-users@lists.freeswitch.org
> Sent: Thu, December 17, 2009 8:03:46 AM
> Subject: Re: [Freeswitch-users] SIP Re-invite
>
> are you doing this trace from the freeswitch box itself?
>
> Mike
>
> On
What is your dialplan on the secondary box?
On Dec 18, 2009, at 9:08 AM, Brian wrote:
I’ve got FS running on a 64 bit OS, and here is more info on the tes
t procedure.
I’ve got one server (primary) that hosts the speaker call (this is m
eant to be a primary conference with a few speakers,
test – more than doubling FreeSWITCH
> capacity in this case. Again, maybe there is something on the FreeSWITCH side
> that I’m doing wrong, but I don’t see what it could be.
>
> Brian.
>
>
> From: Michael Jerris [mailto:m...@jerris.com]
> Sent: Thursday, Dec
On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote:
> On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote:
>> Hi,
>>
>> I'm probably going to regret this - I'm not sure that I'll be able to do
>> this without a lot of pain (nothing to do with FS - more my lack of ability
>> with Vi
I have not seen anyone mention it.
Mike
On Dec 17, 2009, at 11:07 AM, Yehavi Bourvine wrote:
> I'll rephrase my question: Has anyone done that, or should I dig into it?
> After all, Polycom is quite common...
>
> Thanks, __Yehavi:
>
> 2009/12/1
are you doing this trace from the freeswitch box itself?
Mike
On Dec 17, 2009, at 10:48 AM, DJB wrote:
> Anthony,
>
> I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>
> Thank you.
_
k things out.
>
> In case this wasn't apparent I am trying to install FS from trunk.
>
> Thanks,
> Neil
>
> On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris wrote:
> strange, can someone file a bug on this on jira.freeswitch.org and contact me
> off list with ssh
if you don't see it in sofia siptrace but do see it in tcpdump capture then
something very ugly is going on. Either sofia has hung up completely and is
not listening on that port anymore (can other calls go through?) or the packet
you see in tcpdump is not really going to the right port. Can y
Its software, anything is possible with enough time and effort.
Mike
On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote:
> After some discussions with Polycom support it seems that their conferencing
> support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the
> latest and is
I would be curious what the same tests produce with svn trunk of FreeSWITCH.
Mike
On Dec 16, 2009, at 4:49 PM, Brian wrote:
> Hi,
>
> I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to
> see if it will scale better that other solutions. My scenario is to have one
> sp
On Dec 16, 2009, at 9:01 AM, Nameer Kazzaz wrote:
> Hi all,
>Can I set xml_rpc server to run on a specific interface I can set
> the port but not the ip address to bind to. I have a linux server with
> more then one interface. I don't want to use iptables to block it.
No, but you always ha
vorbis, and their dev packages installed.
>
> I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed
> in the dependency lib list. Is this related?
>
> -Neil
>
> On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote:
> looks like ogg de
the typo in your condition.
Mike
On Dec 15, 2009, at 9:27 PM, bcxml wrote:
>
>
> Here is the link to the debug log
>
> http://pastebin.freeswitch.org/11521
>
>
> Brian
>
>
> mercutioviz wrote:
>>
>> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jer
Try turning on debug logs, but from this it looks like its not matching any
extensions.
Mike
On Dec 15, 2009, at 3:11 PM, bcxml wrote:
>
> I have Freeswitch and Microsoft Speech Server 2007 on the same box
>
> When Speech Server initiates a call, I get a sip error message 480
>
> Here is the
You can do that with phrase macros.
Mike
On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote:
> Hello, I create one WAV file that has:
>
> Question + Option 1 + Option 2 + Option 3 + …
>
> I noticed towards end of the file Cepstral Allison starts chopping and
> speeding up.
>
> So my quest
try just 1 video codec in freeswitch codec prefs and make sure you are using
trunk, we fixed quite a few video issues recently.
Mike
On Dec 15, 2009, at 12:54 PM, Jerry Richards wrote:
> I am trying to bring up a video call, but not having much luck. We are only
> getting one-way video (i.e. t
We have not published costs yet, but expect it to be inline with other similar
offerings. I expect the module will initially be available for linux and we
will add other platforms as demand shows a need for it and I can get build
servers up that will be used to produce the binaries. Windows wi
The issue is the demo ivr does not use phrase macros. The line in ru.xml is
for the phrase macros. We should probably change this in the future.
Mike
On Dec 15, 2009, at 5:58 AM, Dmitry Bely wrote:
> On Tue, Dec 15, 2009 at 12:15 AM, Michael Collins wrote:
>>
>>
>> On Mon, Dec 14, 2009 at
That should work fine.
Mike
On Dec 12, 2009, at 2:08 AM, Surajee Ratnayake wrote:
> Hello..
> just want to get the following clarified from the friends in the same domain,
>
> since freeswitch is allowing multiple gtalk user registrations with gtalk
> servers, assume we route gtalk voice calls
A good example of how to use this code would be in mod_rss or mod_voicemail in
tree. I would say look at the doxygen at
http://docs.freeswitch.org/group__switch__ivr.html but it appears that page is
completely broken. I will try to take a look and figure out why this weekend,
in the meantime,
a. However, that
> fails to compile, I presume because of some missing dependency or
> requirement. Is there any tool to tell me what is needed in order to
> build a module ?
>
> Julian
>
> 2009/12/11 Michael Jerris :
>> It just so happens I was looking at this same bug last
It just so happens I was looking at this same bug last night and having
troubles chasing down a solution, if anyone comes up with anything good please
let me know. The basics of this is that automake continues on to other subdirs
if build in one subdir fails.
Mike
p..s. a note on the blog, I
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code
On Dec 11, 2009, at 11:25 AM, Kendall Stauffer wrote:
> Yes, I can do that , I don’t see where I download the source, Sorry to bug
> you.
___
FreeSWITCH-users mailing list
Fr
As i said multiple times on irc last night, we need to see debug logs with sip
trace to see what is going on.
Mike
On Dec 11, 2009, at 11:44 AM, Chris Chen wrote:
> Thanks Frank for sharing your experience. This is the behavior change just
> starting within three days, maybe because of some co
As a note, we are pretty aggressive about making sure all this stuff works
right out of svn without any patches so it should be easy to port freeswitch to
most platforms now.
Mike
On Dec 10, 2009, at 8:57 PM, Brian May wrote:
> On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote:
>> Lack
we also support natpmp and static ip setting.
Mike
On Dec 10, 2009, at 12:21 PM, Fred-145 wrote:
>
> Thanks for the clarification. So it's either UPnP or STUN/port-mapping.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http
I think I fixed the spandsp cross compile issues tonight, but I suspect there
is a good chance that I broke other builds in the process. I also did a bunch
of work to make the OS X Snow Leopard build cleaner today. Testing would be
much appreciated on both.
Mike
On Dec 9, 2009, at 10:47 PM,
sers-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
> Jerris
> Sent: 08 December 2009 16:16
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] no hangup on B leg
>
> We will really need debug logs and sip trac
I recall implementing that back when we released openzap, it should be in there
unless someone chopped it out for some reason. Look for
"zap_channel_send_fsk_data"
Mike
On Dec 9, 2009, at 6:01 AM, François Legal wrote:
> I'm still working on this issue, and decided to take a look at the openz
Our plan for 1.0.5 is that we will also have rpm and deb packages for many
distros on our own repo. Stay tuned. This has been another major reason for
the delay in 1.0.5.
Mike
On Dec 8, 2009, at 1:26 PM, Joseph L. Casale wrote:
>> For those interested, here's how to compile and install Dahdi
The best way to solve this is probably to share the db for presence and
registration between those boxes. If you take a look at the default configs
the settings should be commented there.
Mike
On Dec 8, 2009, at 11:02 AM, Peter P GMX wrote:
> Hello,
>
> is there a way to manually force a pre
I changed the name of key to ikey in trunk.
Mike
> Changing the core db into a MySQL via ODBC caused some problems even after it
> seemed to work. For instance, console help caused an error with an error
> description indicating that a SQL SELECT query including the reserved word
> key has bee
If you can off list provide me with remote login information to this box I can
troubleshot the issue.
Mike
On Dec 8, 2009, at 4:09 AM, Jingwei Yang wrote:
> Hi João, thanks for the reply. But I don't quite get you.. Could you please
> elaborate a little bit? I tried installing libtiff and upgr
We will really need debug logs and sip traces to be able to figure out what
exactly is going on here.
Mike
On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:
> Sorry no, apart from the fact that I was seeing the hangup.
>
>
> I’m wondering if this a bandwidth congestion issue. Is there anywa
Please re-test this with svn trunk of freeswitch and if it is still the case
open up a bug on jira.freeswitch.org in the build system catagory assigned to
me and attach the config.log and config.status from the libs/esl dir to the bug.
Mike
On Dec 7, 2009, at 1:34 PM, Kendall Stauffer wrote:
>
etting because one of my gateways sends INVITEs
> without SDP. I will try to update to the latest trunk today and capture
> traces as Anthony described. If I can't do it today, it might be at the end
> of the week.
>
> Best Regards,
> Jerry
>
>
> From: Michael J
You definitely need to use the settings in combination for what you are trying
to do. Can you explain a bit more what you want to do in what conditions and
maybe we can suggest how to accomplish this. NORMAL_CLEARING is not a failure,
so it can continue on after the bridge unless you specify o
of the codec.
Regards,
Kevin Green
JohnnyVoIP
http://www.johnnyvoip.com
On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris
wrote:
That would binary only, not 64 bit Linux .
On Dec 8, 2009, at 9:17 AM, Kevin Green wrote:
It seems you can get a copy of either the binaries or the source
That would binary only, not 64 bit Linux .
On Dec 8, 2009, at 9:17 AM, Kevin Green wrote:
It seems you can get a copy of either the binaries or the source by
doing the following:
Review & execute SILK Agreement - attached. NOTE - please add your
Skype login to this form also.
Return execu
Also I have seen some people reporting that the new tickless timers in newer
kernels work better. You may want to try those.
Mike
On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote:
> Did you do each thing alone too to tell the difference?
> -hp alone, disable monotonic alone (i did not see
Please report bugs to jira.freeswitch.org.
Mike
On Dec 6, 2009, at 11:45 PM, Seven Du wrote:
> Hi,
>
> I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
> to trunk code, no sound after att_xfer.
>
> Then I rebuild FS 15807 with a fresh checkout, but still using the old
> c
This bug has been now closed out in jira due to no response for requested
information. If you wish to resolve this issue please follow up on your bugs
when information is requested.
Mike
On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote:
>
> Nope, I wanted to make sure that this is inde
looking to set
this up without needing proxies I would want to use srv records and naptr
records and a provider that would balance using these including failiover.
Mike
On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote:
> On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote:
>> The easi
:
>>>>>
>>>>> As I see it, switch_cond_next() currently is just a do_sleep(1000).
>>> Yes,
>>>>> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND"
>>>>> overrides that.
>>>>>
>
what revision were you at prior to upgrade or can you narrow the range of
versions that broke this any more (or even better the exact version that broke
this). Please post this bug to http://jira.freeswitch.org.
Mike
On Dec 3, 2009, at 10:30 AM, Milena wrote:
> Hello,
>
> It was all ok until
with the right clients, it nearly always works well. with a client that does
not support stun or at least rfc 3581 the results are much more sketchy and
require more hacks on the server side, but with enough effort can almost always
be made to work.
Mike
On Dec 3, 2009, at 7:17 AM, Fred-145 w
You may want to try this again with latest svn trunk. We have done quite a lot
of work to make nat support much better sense 1.0.4
Mike
p.s. I can't comment about version 1.4 due to broken flux capacitor.
On Dec 3, 2009, at 4:36 AM, Henry Huang wrote:
> My freeswitch is using public IP. I s
You can turn up the full freeswitch debug or enable the siptrace on the sip
profile to get more information about this. This looks like a nat related
issue getting no response from the provider. A sip trace is probably the best
tool to figure this one out.
sofia profile internal siptrace on
The easiest place to do this is at the point you send the calls to FreeSWITCH.
How are the calls coming in?
Mike
On Dec 2, 2009, at 7:49 PM, Tim Uckun wrote:
> I have read some of the archived emails about HA, loadbalancing,
> failover etc and I am still a bit confused about how I could set up
The behavior is probably expected, the unhelpful error is probably undesirable
but it would make a mess of the dial-plan to clean that up.
Mike
On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote:
> Is this reasonable given it was the only call in FreeSwitch at the time? How
> can this situation be corr
http://wiki.freeswitch.org/wiki/Mod_commands#originate
Usage: originate |&()
[] [] [] [] []
You can do this via shelling out to fs_cli like your example below or using esl
directly from php:
http://wiki.freeswitch.org/wiki/Esl
Mike
On Dec 2, 2009, at 1:23 PM, eaf wrote:
>
> I need a wa
You could also use the scheduler to run the jsrun command inside FreeSWITCH.
Mike
On Dec 3, 2009, at 8:31 AM, Rob Forman wrote:
> What about cron?
>
> Create a cron entry like:
> */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()"
>
> But if you're just dumping global va
acing thread.
> Which would be dead-quiet at idle time, but, of course, would stop scaling
> at some point due to excessive number of system calls.
>
> Thanks.
>
>
> Michael Jerris wrote:
>>
>> In short. No, you can not for many reasons. The milisecond tic is
&g
In short. No, you can not for many reasons. The milisecond tic is
used throughout the code even when there is not any calls up. You can
grep for switch_cond_next if you would like to see where but it is
required to keep our global timestamp and for pacing the scheduler
among other service
This is keeping track of a place in the music on hold so your hold
music does not start back up at the same place every time. If you
don't want to do this it is a module that you don't need to load and
you can get your moh from any soundfile at your choice in configuration.
Mike
On Dec 2,
I think I just fixed this a few minutes ago, it is running test builds on the
build servers now to verify.
On Dec 1, 2009, at 2:19 PM, John Platts wrote:
>
> I attempted to do a make current with revision 15739, but some of the Sofia
> source files will not compile with revision 15739. Those
The only way this would happen would be if this is set to proxy media
not bypass. Are you setting both?
Mike
On Dec 1, 2009, at 10:08 AM, Juan Backson wrote:
In the following trace,102 is FS IP, 104 is calling party and 13
is called party.
with bypass_media, FS still changesc=IN
What is the jira bug number on this voicemail email issue? I don't
recall seeing it.
Mike
On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine
wrote:
> > Are you on SVN trunk? As far as I recall the callee_id_number/name
> stuff isnt in 1.0.4.
>
> No, because the SVN has problems with Emailing th
make openzap is the correct way to build when using with openzap/freeswitch.
If you are having issues with this you should check with sangoma support as to
why that build of the drivers is not supporting it properly and what version
you should be using.
Mike
On Nov 30, 2009, at 5:41 AM, Franç
It depends on the timing of when your increment and decrement are vs when the
sql calls to push the events into the tables that are used for show calls are.
Also, the sql calls are batched and queued causing a little delay (less than a
second). If your doing a lot of short lived calls there is
he profile
>> so it can find it.
>>
>> /b
>>
>> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
>>
>>
>>> Try an alias on the sip profile.
>>>
>>> Mike
>>>
___
F
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html
MySQL Connector/ODBC now supports batched statements. In order to enable
cached statement support you must switch enable the batched
statement option (FLAG_MULTI_STATEMENTS,
67108864, or Allow multiple
Of course. Please read through the default configs and the getting started
guide and xml dialplan information on the wiki.
Mike
On Nov 26, 2009, at 12:38 PM, Orien Love wrote:
> Is there any way to build a dial plan so that when an extension calls
> itself the call is automatically put to that
In this case you should not need 2 profiles either.
On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote:
> It's a windowsserver which is behind a router.
>
> Which profile should local-network-acl be specified on?
>
> When I bridge calls to the outside world, should I use
> sofia/internal/@gatewa
Your using outbound socket and you hangup the call, so it tells you it
is done with the server disconnected message and drops the
connection. This is all as expected. I guess I don't understand what
you think is the problem. This code is doing exactly what I would
expect it to do.
Mike
There are execute_complete events. I can't recall everything that is in them
but they should always be fired.
Mike
On Nov 25, 2009, at 8:38 PM, Josh Rivers wrote:
> Is there a way of determining if a call-command sent to a session via ESL has
> completed? Is there a return event which is alw
from
http://svn.freeswitch.org/svn/freeswitch/trunk/conf/sip_profiles/internal.xml
It appears this never made the wiki, could someone please get it on there.
Thanks
Mike
On Nov 25, 2009, at 6:21 PM, John Platts wrote:
>
> How do I turn on dialplan processing of 302 responses? I can solve m
On Nov 25, 2009, at 5:18 PM, Adam Ford wrote:
> Samuel,
>
> FreeSWITCH has a Skype module that uses Skype client instances to connect to
> the Skype network, you can read about it at
> http://wiki.freeswitch.org/wiki/Skypiax
>
> As far as an official Skype module for non-Asterisk PBX-es, it loo
"something that is not available in that lib at this time."
Mike
On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote:
> can please tell me how can i exchange session state into sip library.
>
> Thanks
> srinivas
>
> On Wed, Nov 25, 2009 at 11:47 PM, Michael J
For that you would need to fully exchange session state into the sip library,
something that is not available in that lib at this time.
On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote:
> HI,
> thanks for your reply, my requirement is i am doing failover stuff with
> freeswitch. i dont wa
You can send the call with secure enabled and if it supports it it will use it.
Mike
On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote:
> Hello,
>
> We have a mix of phones that support RTP encryption and those that do not.
> I have to support both types in the meanwhile, and would like to
It's possible it does not. I just added some code to set it on auto-adjust so
it might be there sometimes now. You might need to add some code in mod_sofia
to add it other times. Maybe it makes sense to move that var setting down to
switch_rtp.c. Patches for this would be welcome.
Thanks
M
In trunk there is a sofia profile setting to allow dialplan processing of 302
responses. This won't get you back into your same javascript, but you can
probably do something clever from there.
Mike
On Nov 24, 2009, at 5:04 PM, John Platts wrote:
>
> I have considered writing JavaScript code
Its a feature we don't have, patches welcome.
Mike
On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote:
> Hi members,
> I’m controlling freeswitch with the conference module via xmlrpc.
>
> Is it desired that the kick command can only kick users that are connected to
> the conference?
> Is t
FreeSWITCH will kill the calls when you shut it down, if you intentionally kill
the network without shutting down FreeSWITCH the only thing you can do is
enable session timers or rtp timers in the soft phones to kill the call when
FreeSWITCH dies or when the call is over.
Mike
On Nov 25, 2009,
sent signal r_respond
>>nua(0x8fc3d0): recv signal r_respond 200 OK
>>nua: nua_handle_destroy: entering
>>nua: nua_stack_set_params: entering
>>nua(0x8fc3d0): sent signal r_destroy
>>nua: nua_handle_magic: entering
>>nua: nua_handle_destroy:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide
http://wiki.freeswitch.org/wiki/Interop_List
On Nov 25, 2009, at 1:36 AM, ovvenkat wrote:
> Hi .
>
> Could you please tell me, How to connect sip phone (which one is more
> friendly with freeswitch) to freeswitch. How I can check whether
"you should use execute_complete events to tell when a command you tried to
execute has finished and not poll the channel for a variable to be set because
FreeSWITCH is an asynchronous application in the mode you are describing and
you can never be sure of the timing."
You are STILL polling for
On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote:
> Hi,
>
> I'm trying to setup call transfer for a phone without a transfer button. I
> was on IRC last night and got some pointers to how this is setup in
> dialplan.xml and features.xml and what "bind meta app" does.
>
> Once it became cle
1. can you supply a trace of this esl communications.
2. is it inband or rfc2833 dtmf ?
MIke
On Nov 24, 2009, at 3:59 AM, velusamy velu wrote:
> Yes, I am using async mode only..
>
> On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris wrote:
> async?
>
> On Nov 24, 2009, at
What does this have to do with uniMRCP?
Mike
On Nov 24, 2009, at 9:54 AM, Imthiyaz Ahmed wrote:
> Hi
>
> Can we enable passive recording in freeswitch ,wanpipe ,openzap , we
> are using a sangoma tapping system with freeswitch.
___
FreeSWITCH-user
async?
On Nov 24, 2009, at 2:22 AM, velusamy velu wrote:
> Dear All,
> I am using Perl ESL::IVR module to develop a simple IVR. I have
> filtered DTMF events. I have also set playback_terminators to cut the
> playback when giving the digits. I have faced problem that DTMF event has no
Your not telling anything to call your callback.
On Nov 24, 2009, at 1:03 AM, Baskar wrote:
> Hi,
>
> I want to check value given to the javascript with conditions whether it is
> voicefile, extension or mobile Number when i press the dtmf value.
>
> Steps i need to check in javascript:
>
>
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