You need to tack on {ignore_early_media=true}sofia/profile/
[EMAIL PROTECTED] or {ignore_early_media=true}sofia/gateway/provider.com/
number
Either way that makes it not return and ignore early media till the
remote side answers and won't start playing till then.
/b
On Jul 23, 2008, at 8:07
> you can list dial strings separated buy a | (pipe) symbol to do successive
> calling and , (comma) to do simultaneous or a combo of both
thank you very much, it works! I used a | symbol and variable "call_timeout".
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Hi,
We see originate function as follows:
result = new_session.originate(session, dest[[, dialplan],
context], cid_name], cid_num], network_addr], ani], aniii], rdnis],
username], to]);
Where to= timeout
When does the timeout counter start, after call to the originate Api, or
after the
mod_snom has more work to do the complete management of the list of
users and button states. Also have to fix the action URL's that need
auth.
Seriously you can do true key system emulation with this and the
action URL's.
/b
On Jul 23, 2008, at 9:16 PM, Brian Snipes wrote:
> Can the use
Yah I sent it in to christian. Fun thing is the guy that writes the
firmware will be at Cluecon :P
/b
On Jul 23, 2008, at 9:13 PM, Michael Jerris wrote:
> They just ran into the issues this afternoon, I don't think we have a
> report into snom yet
>
> Mike
__
В сообщении от Thursday 24 July 2008 02:35:58 Star Wind написал(а):
> > This might be what you want.
> >
> > bridge data="sofia/blah/user%domain|sofia/blah/user2%domain| ..."
>
> thank you, but how I can manage timeouts?
Вроде так:
May be that variable?
--
С уважением,
Кривушин Михаил Евгенье
Can the userlist be 2 columns? This might be an option until I can get the
existing sidecars I have to work the way I want them to.
Brian
On Wednesday 23 July 2008 9:01:11 pm unknown wrote:
> If you don't mind I would suggest the operator console using softphone such
> as Bria 2.2 which you can
They just ran into the issues this afternoon, I don't think we have a
report into snom yet
Mike
On Jul 23, 2008, at 10:02 PM, Brian Snipes <[EMAIL PROTECTED]> wrote:
> Do you know if any of the snom firmware works as it should for that
> functionality or has snom given any kind of estimate on
well it'll work but there are a few issues related to it... i'll work
it out shortly... all that functionality would live in mod_snom.
/b
On Jul 23, 2008, at 9:02 PM, Brian Snipes wrote:
> Do you know if any of the snom firmware works as it should for that
> functionality or has snom given any
Do you know if any of the snom firmware works as it should for that
functionality or has snom given any kind of estimate on when we can expect a
version of firmware that will work properly with it? :-)
Brian
On Wednesday 23 July 2008 8:52:33 pm Brian West wrote:
> Thats why snom has come up wit
If you don't mind I would suggest the operator console using softphone such
as Bria 2.2 which you can add all sip contacts into its contacts list, where
you can monitor all extensions with detailed information such as "ringing"
"talk 3001" "available" etc, when working with Freeswitch. The cost is
Hi ,
I have started using freeswitch and impressed with it, how easy to set up and
run quick examples. It is awesome!
What I do here is I bridge the call between IVR and a remote phone. Then remot
eparty pick up the phone and, IVR plays a prompt. I want the remote user
listen all prompt conte
Thats why snom has come up with this buttons stuff I was telling you
about. This is the exact issue that solves. The ground work for
supporting this is already there in mod_snom but we found some bugs in
the snom firmware related to that. So you could expect complete key
emulation with a
There should be no performance issues with -nf. This is the exact
reason it exists.
Mike
On Jul 23, 2008, at 9:43 PM, "John Skopis (Lists)" <[EMAIL PROTECTED]>
wrote:
> Birgit Arkesteijn wrote:
>> Hi all,
>>
>> We've got an older version of FreeSWITCH (Trunk 7948) running on a
>> Linux
>
Birgit Arkesteijn wrote:
> Hi all,
>
> We've got an older version of FreeSWITCH (Trunk 7948) running on a Linux
> x86_64 machine. At the moment it's crashing few times a day, making our
> services very unreliable.
>
> At the moment we don't have the time to rebuild this version, so I'm
> looki
Ok.. I need some suggestions. Currently I have a couple of snom 370 phones
with 2 sidecars on each phone for the receptionist and her backup location.
Those phones monitor the status of ~55 extensions. The Snoms seem to be
having a devil of a time doing this. Several times a day they get RE
Guys,
I was just looking over the source directory an I noticed that the
common files for an open source project may need some serious attention:
AUTHORS
INSTALL
NEWS
README
I'm sure that the COPYING file is just fine.
NEWS and README are completely empty. Should we maybe just put
Thank you.
I have followed the instructions of this link but FS can't find the user.
I've stopped IPtables and have forwarded the port 5090 to the box.
sofia status
=
internal
The default sample rate is 8k which is as small as it gets.
mod_shout would let you record mp3s or you could use socks to change the
codec inside the wav to something smaller but that's about it.
On Wed, Jul 23, 2008 at 4:01 PM, Michael Collins <[EMAIL PROTECTED]>
wrote:
> I think he means, "I
I think he means, "I don't need CD-quality audio or anything close to
that - I just want a lower sampling rate so that my wav file isn't so
big."
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony Minessale
Sent: Wednesday, July 23, 2
Anthony,
Thanks for the heads up on that. This is important because not all CDRs
reflect a 2-legged call. Also, don't forget that you can set your CDRs
to record A leg, B leg, or both legs.
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Ohh..thats makes the work easier. Thanks for the tip anthony!
Regards,
ashutosh
On Wed, Jul 23, 2008 at 8:38 PM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> on A legs you have b_leg_uuid and on B legs you have a_leg_uuid to get the
> uuid of the other half of the call.
>
>
>
> On Wed, Jul 23
nice job. Thank you.
On Wed, Jul 23, 2008 at 3:35 PM, UV <[EMAIL PROTECTED]> wrote:
> Mod_shout for windows is ready:
> http://jira.freeswitch.org/browse/FSBUILD-63
>
>
>
>
>
>
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswit
on A legs you have b_leg_uuid and on B legs you have a_leg_uuid to get the
uuid of the other half of the call.
On Wed, Jul 23, 2008 at 3:09 PM, Ashutosh <[EMAIL PROTECTED]> wrote:
> ok, is there any way or a common variable through which i can relate calls
> of the same id. What does the field s
Well we are only as good as our community, so without your feedback we are
nothing so keep it up!
On Wed, Jul 23, 2008 at 3:07 PM, Ashutosh <[EMAIL PROTECTED]> wrote:
> Yeah, That was implied in the very first line "FS Team rocks!"
>
> Cheers again all!!
>
> -ash
>
> On Wed, Jul 23, 2008 at 7:56
Mod_shout for windows is ready: http://jira.freeswitch.org/browse/FSBUILD-63
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UNSUBSCRIBE:http://lists.freeswitc
what do you mean by "more information"?
On Wed, Jul 23, 2008 at 9:48 AM, Frank - IMPACT <[EMAIL PROTECTED]> wrote:
> Running FreeSwitch Version 1.0.trunk (9111) on RH 8
>
> Record_session works fine with this
>
>
>
>
> data="$${base_dir}/recordings/${strftime(%Y-%m-
ok, is there any way or a common variable through which i can relate calls
of the same id. What does the field sip_call_id field do ?
Best regards,
ashutosh
On Wed, Jul 23, 2008 at 7:57 PM, Brian West <[EMAIL PROTECTED]> wrote:
> no each leg has its own uuid.
>
> /b
>
> On Jul 23, 2008, at 2:41
Yeah, That was implied in the very first line "FS Team rocks!"
Cheers again all!!
-ash
On Wed, Jul 23, 2008 at 7:56 PM, Gonzalo Servat <[EMAIL PROTECTED]> wrote:
> On Wed, Jul 23, 2008 at 4:34 PM, Ashutosh <[EMAIL PROTECTED]>
> wrote:
>
>>
>> [..snip..]
>>
>> Salutes to the never ending spirit
Please follow http://wiki.freeswitch.org/wiki/Example_Offsite_phones
I managed to get the SIP remote extensions work for the double NATed
environment. Just need to be more careful.
Thanks,
Chris
On Wed, Jul 23, 2008 at 3:59 PM, Jair Santos <[EMAIL PROTECTED]> wrote:
> Yes.
>
> And I want to regi
Yes.
And I want to register this user to the external IP . I am forwarding the
port to the box. How can I register the user trough the ext IP ?
Jair Santos
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Brian West
> Sent: Wednesday, Jul
no each leg has its own uuid.
/b
On Jul 23, 2008, at 2:41 PM, Ashutosh wrote:
> Hi Michael,
>Will two legs of the same call show up the same uuid in the cdr ?
Brian West
sip:[EMAIL PROTECTED]
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Freeswitch-users@
On Wed, Jul 23, 2008 at 4:34 PM, Ashutosh <[EMAIL PROTECTED]> wrote:
>
> [..snip..]
>
> Salutes to the never ending spirit of man, thats called Anthony!!
>
.. and the entire FS team!
- Gonzalo
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Freeswitch-users@lists.fr
Your domain is 192.168.1.117 ... Is freeswitch behind nat?
/b
On Jul 23, 2008, at 2:21 PM, Jair Santos wrote:
> API CALL [sofia(status)] output:
> Name
> Type Data State
> =
> =
> =
> =
> =
> =
> =
> =
> =
> =
> =
> =
Hi Michael,
Will two legs of the same call show up the same uuid in the cdr ?
Thanks,
ashutosh
On Wed, Jul 23, 2008 at 7:37 PM, Michael Collins <[EMAIL PROTECTED]>
wrote:
> Is there any reason that the uuid field doesn't work for you?
>
>
> --
>
> *From:* [EMAIL
Is there any reason that the uuid field doesn't work for you?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ashutosh
Sent: Wednesday, July 23, 2008 12:18 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Equivalent of ast
> This might be what you want.
>
> bridge data="sofia/blah/user%domain|sofia/blah/user2%domain| ..."
thank you, but how I can manage timeouts?
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/
> you can list dial strings separated buy a | (pipe) symbol to do successive
> calling and , (comma) to do simultaneous or a combo of both
thank you VERY much, but tell me please, how can I set up timeout for calling?
for example:
- 20 seconds (or 3 rings) - - 20 seconds... ?
and what will h
Anthony rocks, FS team rocks, will keep rocking!!
3 hour release show a good practice of agile programming at its best.
Essentially, asterisk was not bad; actually, the fact is that FS is so so
much outstanding, cleaner and working ,that asterisk pales out.
Frankly, after seeing this post thread,
API CALL [sofia(status)] output:
Name Type Data
State
=
internal profile sip:[EMAIL PROTECTED]:5060
RUNNING (0)
Hi,
I have been looking for an attribute in the FS CDR which will be unique
for a call, like we had uniqueid field in asterisk cdr. Someone have any
idea ?
Thanks,
ashutosh
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http:/
You don't add domain name in there like that.
show me the output of "sofia status".
/b
On Jul 23, 2008, at 2:11 PM, Jair Santos wrote:
Brian West
sip:[EMAIL PROTECTED]
___
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Freeswitch-users@lists.freeswitch.org
ht
I have changed the user 1001 to
I can see the 1001 user registered in the phone.
Trying to
phone->FS->nat->phone
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
West
S
It's fairly obvious you don't have a user 1001 in domain 24.67.78.200,
I also recommend you can update to the latest code as of now.
Did you modify the default config? And what scenario are you trying
to make this work with?
On Jul 23, 2008, at 1:53 PM, Jair Santos wrote:
2008-07-23 11
I added to the top.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
West
Sent: Wednesday, July 23, 2008 11:45 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] registered user calling
Oh btw press F8 if you're on
Changing to
[EMAIL PROTECTED]
2008-07-23 11:49:10 [WARNING] mod_sofia.c:1807 sofia_outgoing_channel()
Cannot locate registered user [EMAIL PROTECTED]
But the user is registered as I can see in the phone.
Jair Santos
-Original Message-
From: [EMAIL PROTECTED]
[mail
Oh btw press F8 if you're on linux. The default config has 1000 thru
1019 configured already so if you added that at the bottom of default
then the Local_Extensions is overriding it.
/b
On Jul 23, 2008, at 1:33 PM, Jair Santos wrote:
Brian West
sip:[EMAIL PROTECTED]
Users don't register to the nat profile at all and SHOULDNOT. The
default config makes the assumption that all users are registering to
the internal profile. Anything outside of that would require you to
know a bit more on how FreeSWITCH works. Join us on IRC #freeswitch
on irc.freenode.
This should get you started.
http://wiki.freeswitch.org/wiki/Tls
/b
On Jul 23, 2008, at 1:29 PM, Peter P GMX wrote:
> Hello,
>
> has anybody managed to setup TLS? When I change tls to "true" in
> internal.xml, then freeswitch doens't listen on any ports (5060 5061).
> I use freeswitch 1.0.0-0u
In the default context I have
[EMAIL PROTECTED]
2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
features
2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443
switch_ivr_bind_dtmf_meta_session() Bound B-Leg:
Only 59 diggs? I figured something like this would be of more interest to
the community at large
> From: Brian West <[EMAIL PROTECTED]>
> Reply-To:
> Date: Tue, 22 Jul 2008 11:56:54 -0500
> To:
> Cc: "[EMAIL PROTECTED]"
> <[EMAIL PROTECTED]>
> Subject: [Freeswitch-users] FreeSWITCH in the press
Hello,
has anybody managed to setup TLS? When I change tls to "true" in
internal.xml, then freeswitch doens't listen on any ports (5060 5061).
I use freeswitch 1.0.0-0ubuntu1~ppa4
Is there any tutorial available (could not find it while googling)? I
would like to set it up with Snom phones (Sn
Hi Anthony,
To be honest, I didn't expect changes to lightly be made which broke
backwards compatibility in something which is notionally in beta, but I
didn't (and don't) complain about it - it's just something which might
have caused Birgit a bit of grief if she'd followed your advice.
Loo
What else would you expect?
That is the risk you take using the beta version of open source free
software.
Since we are probably tagging 1.0.1 today TRUNK and forward is the only
version we are going to support with our usual triple-your-money-back
guarantee!
On Wed, Jul 23, 2008 at 9:37 AM, Da
Running FreeSwitch Version 1.0.trunk (9111) on RH 8
Record_session works fine with this
But the sample rate and resulting wav file are way more information than I
need. I tried to half the sample rate with this inserted before we run
record_session.
With respect, this may well not be true. I've had a handful of "upgrade
breaks things"
issues where, for example, variables in CDRs change name, or dtmf-digit
becomes dtmf-
string (or the other way round - can't remember) in an event, etc.
--Dave
if you would have typed "make current" when you
if you would have typed "make current" when you sent the original you would
be done by now.
On Wed, Jul 23, 2008 at 8:52 AM, Birgit Arkesteijn <[EMAIL PROTECTED]>
wrote:
> Hi Anthony and Brian,
>
> Thanks for your replies!
>
> I know we should upgrade and we will at some point, but honestly (cro
Hi Anthony and Brian,
Thanks for your replies!
I know we should upgrade and we will at some point, but honestly (cross
my heart and hope to die), I just don't have the time at the moment.
Thanks, Birgit
On 23/07/08 14:17, Anthony Minessale wrote:
> you do realize we are on 9139 now right? you
you will need event socket for what you want to do.
On Wed, Jul 23, 2008 at 4:13 AM, Boris Krivonog <[EMAIL PROTECTED]>
wrote:
> That was fast :)
>
> I'm struggling with the FreeSwitch documentation available on the net and
> cannot tie all parts together to see the big picture for what I'm try
you do realize we are on 9139 now right? you would be missing out not to
update.
On Wed, Jul 23, 2008 at 8:05 AM, Brian West <[EMAIL PROTECTED]> wrote:
> http://jira.freeswitch.org/browse/MODAPP-89
>
> I'll recommend that you update. I'll bet its a memory related crash
> or an assert in sofia.
you can list dial strings separated buy a | (pipe) symbol to do successive
calling and , (comma) to do simultaneous or a combo of both
assume A B C and D are all valid FS dial strings such as sofia/default/
[EMAIL PROTECTED]
you can say
A|B|C|D call A then B then C then D till someone answers
You might be better off using Mod_event_socket and some lua. Check
out mod_lua and mod_event_socket on the wiki.
/b
On Jul 23, 2008, at 4:13 AM, Boris Krivonog wrote:
That was fast :)
I'm struggling with the FreeSwitch documentation available on the
net and cannot tie all parts together
http://jira.freeswitch.org/browse/MODAPP-89
I'll recommend that you update. I'll bet its a memory related crash
or an assert in sofia.
/b
On Jul 23, 2008, at 4:56 AM, Birgit Arkesteijn wrote:
> Hi all,
>
> We've got an older version of FreeSWITCH (Trunk 7948) running on a
> Linux
> x86_64
This might be what you want.
bridge data="sofia/blah/user%domain|sofia/blah/user2%domain| ..."
/b
On Jul 23, 2008, at 12:15 AM, Star Wind wrote:
> Who can tell me, how I can create a static ring group? I'm using
> third-party SBC, which changes "Contact" field on every
> registration, every
Hi all,
We've got an older version of FreeSWITCH (Trunk 7948) running on a Linux
x86_64 machine. At the moment it's crashing few times a day, making our
services very unreliable.
At the moment we don't have the time to rebuild this version, so I'm
looking for an equivalent of the "safe_asteris
That was fast :)
I'm struggling with the FreeSwitch documentation available on the net and
cannot tie all parts together to see the big picture for what I'm trying to
achieve:
>From an external application drive operations like:
* originate a call to a call leg
* feed dtmf to call leg
* collect d
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