On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote:
> On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:
>> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
>>>
>>> Well, if you're running multiple machines, waiting for it to drainstop
>>> isn't that big of a deal unless you're in
Hi,
I have some confusion about FreeSWITCH's Mozilla Public License 1.1. I do
understand that me or any one can change provided code according to our
customization needs and we are not bound to share our changes as long as we
are not distributing it, right?
Now, i have been doing R&D on MSN and Y
>> Well, if you're running multiple machines, waiting for it to drainstop
>> isn't that big of a deal unless you're in some sort of hurry, right?
>> Give it an hour or so to drainstop, then kill 'em.
>
>Yes that's exactly what I'm trying to do. The problem is some people will
>only try one IP ad
On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:
Hi,
On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
Well, if you're running multiple machines, waiting for it to
drainstop
isn't that big of a deal unless you're in some sort of hurry, right?
Give it an hour or so to drain
Hi,
On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
>
> Well, if you're running multiple machines, waiting for it to drainstop
> isn't that big of a deal unless you're in some sort of hurry, right?
> Give it an hour or so to drainstop, then kill 'em.
Yes that's exactly what I'm t
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
-- Meet u
snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX- but
delivers XX to FS.
Thanks Brian
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 6:41 PM
To: frees
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
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
Freeswitch-users@lists
snom320-SIP 6.5.17.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 5:40 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match
What firmware?
/b
On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote:
It’s a SNOM 320.
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://list
It's a SNOM 320.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 4:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match
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
br...@freeswitch.org
-- Meet us at ClueCon! http://www.clueco
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
On Thu, Jun 11, 2009 at 6:57 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
> Not sure where enhancement requests should be posted, but here it is
> anywa
If the phone sends them with dashes in them the phone IS BROKEN and
should be smashed with a hammer.
/b
On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote:
The users entering numbers into their phonebooks are able to
recognize the number more easily.
I will tell them to forget it and make the ph
The users entering numbers into their phonebooks are able to recognize the
number more easily.
I will tell them to forget it and make the phone numbers numeric only.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Micha
Will do
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 11 June 2009 22:51
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphan
Not sure where enhancement requests should be posted, but here it is
anyway
I would dearly love to be able to send a status event that returns an
event style output that provides machine readable output rather than the
wordy human readable response. (I hate parsing)
Is there such an even
If they were still showing in status, can you use gcore to dump a core
next time this happens, leave it running somewhere we can get to it
and post a thread apply all bt to Jira.
Mike
On Jun 11, 2009, at 5:40 PM, "Nik Middleton" > wrote:
It was the output from show channels. I’ve rebooted
It was the output from show channels. I've rebooted the server now, so
I can't run show calls. I'll see what happens tomorrow. Certainly
running status showed 6 sessions
All calls are initiated using and 'Originate' from an inbound socket
Regards
___
it may be a race in the sql event handler where the delete comes before the
insert on a really short call.
how many sessions did "status" report were in use?
On Thu, Jun 11, 2009 at 4:07 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
>
>
> Ok, so I did a mere 86,000 calls today, b
back to back user agent! :-)
Thanks! I just ask google!
NOx-WHV wrote:
>
> Thanks for your answer.
>
> Can you just announce b2bua.
>
>
>
> Brian West-3 wrote:
>>
>>
>> On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote:
>>
>>>
>>> Hello Freeswitch User!
>>>
>>> I am using FS since a few week
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.
-MC
Brian West
br...@freeswitch.or
It does if you do a blind transfer... if you're talking attended
transfers thats a whole different ball of wax...
/b
On Jun 11, 2009, at 4:17 PM, John Wehle wrote:
It appears from some limited testing that the original caller id is
always
shown when the call is transfered. Is there some wa
what?
On Jun 11, 2009, at 4:16 PM, NOx-WHV wrote:
Can you just announce b2bua.
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.f
On Thu, Jun 11, 2009 at 2:17 PM, John Wehle wrote:
> > It appears from some limited testing that the original caller id is
> always
> > shown when the call is transfered. Is there some way to have the person
> > making the transfer show up as the caller id?
>
> To answer my own question it appea
On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb wrote:
> I have a match expression for outbound calls as “\d{10}”. It’s fine for
> unformatted numbers. Not knowing any better, I created another extension to
> handle numbers formatted like XXX-XXX-, which is easier to read and
> exists in one hard
On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
>
>
> Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6
> sessions remaining like the one below (number and ISP changed)
>
>
>
> Anyone have an idea why these 6 sessions remain? I
> It appears from some limited testing that the original caller id is always
> shown when the call is transfered. Is there some way to have the person
> making the transfer show up as the caller id?
To answer my own question it appears that the information is available
in the sip_h_Referred-By va
Thanks for your answer.
Can you just announce b2bua.
Brian West-3 wrote:
>
>
> On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote:
>
>>
>> Hello Freeswitch User!
>>
>> I am using FS since a few weeks. My intent is to have clients who
>> uses TLS
>> and SRTP for a full encrypted call.
>>
>> I jus
Ok, so I did a mere 86,000 calls today, but when it was all over, I had
6 sessions remaining like the one below (number and ISP changed)
Anyone have an idea why these 6 sessions remain? I also had 120 calls
that I didn't get a hang-up for, but that might be me not processing the
events fas
Well, if you're running multiple machines, waiting for it to drainstop isn't
that big of a deal unless you're in some sort of hurry, right? Give it an hour
or so to drainstop, then kill 'em.
Would it not be simpler to try to do something with re-invites or REFER,
assuming your endpoints suppor
On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote:
Hello Freeswitch User!
I am using FS since a few weeks. My intent is to have clients who
uses TLS
and SRTP for a full encrypted call.
I just managed it, that calls are encrypted with TLS and SRTP. My
second aim
ist to redirect this calls for r
Hello Freeswitch User!
I am using FS since a few weeks. My intent is to have clients who uses TLS
and SRTP for a full encrypted call.
I just managed it, that calls are encrypted with TLS and SRTP. My second aim
ist to redirect this calls for reduce the processing of the server. I only
use the FS
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
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-user
I have a match expression for outbound calls as "\d{10}". It's fine for
unformatted numbers. Not knowing any better, I created another extension to
handle numbers formatted like XXX-XXX-, which is easier to read and
exists in one hard phone's phonebook.
It looks like: "^1?(\d{3})-(\d{3})-(\
That's exactly what I do.
Between dispatcher and FLAGS/GFLAGS this is easy to do in OpenSIPS/SER.
On Thu, Jun 11, 2009 at 12:54 PM, Anthony
Minessale wrote:
> or you can put a sip proxy in front of 2 boxes where you can control the
> flow of traffic.
> when you want to upgrade one, take all the t
Your syntax is also wrong, it should be
and NOT field=${varname}$
Math
On 11-Jun-09, at 2:07 PM, Brian West wrote:
try destination_number
/b
On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote:
http://pastebin.freeswitch.org/9365
I do not know what I am doing wrong. I am trying to set th
try destination_number
/b
On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote:
http://pastebin.freeswitch.org/9365
I do not know what I am doing wrong. I am trying to set the
effective_caller_id_name and _number depending on the originating
extension.
I tried:
expression="^100
On Jun 11, 2009, at 2:24 PM, John Dalgliesh wrote:
> On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
>> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote:
>>>
>>> Exactly. You probably want to have something like this anyways, so
>>> that
>>> when someone accidentally unp
On Thu, Jun 11, 2009 at 11:24 AM, John Dalgliesh wrote:
> On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
>
>> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote:
>>
>>>
>>> Exactly. You probably want to have something like this anyways, so that
>>> when someone accidentall
Michael,
Removing everything between the tag in
sip_profiles/internal/example.xml did the trick - no error message on FS
startup. I'm running 13723.
2009-06-11 07:21:03.609317 [INFO] switch_event.c:564 Activate Eventing
Engine.
2009-06-11 07:21:03.612274 [DEBUG] switch_event.c:552 Create
On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote:
Exactly. You probably want to have something like this anyways, so that
when someone accidentally unplugs the system, or the disks/CPU/RAM crash,
you’re not stuck.
That is,
I assume he's talking about hardware failures here :P
But to answer the question: crashes are easy to deal with. With a crash
you have lost the calls that are in progress anyway; you don't have to
manage a gradual transition.
Currently, since FS is quite quick to start up, I am just relaunchin
http://pastebin.freeswitch.org/9365
I do not know what I am doing wrong. I am trying to set the
effective_caller_id_name and _number depending on the originating extension.
I tried:
and
and
But each got substituted with the name of the extensi
OK thanks that is what I thought the general way of doing it would be. But
it seems a bit wasteful to have that SIP proxy there the whole time
especially when I am using FS in the role of an SBC.
The problem with the graceful restart of course is that you have to wait
for the calls count to
On Thu, Jun 11, 2009 at 10:43 AM, wrote:
> Haha, good point on the FreeSwitch Solutions site... Suits - very
> professional. =)
> Please don't think I'm telling you guys what to do, I know you don't
> need that. It IS your software and your site, and you've done a HELL of
> a job with it so fa
Haha, good point on the FreeSwitch Solutions site... Suits - very
professional. =)
Please don't think I'm telling you guys what to do, I know you don't
need that. It IS your software and your site, and you've done a HELL of
a job with it so far.
It was just a thought. Sorry if I offended.
On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote:
> Exactly. You probably want to have something like this anyways, so that
> when someone accidentally unplugs the system, or the disks/CPU/RAM crash,
> you’re not stuck.
>
>
>
> That is, until FreeSWITCH can record its internal state to
Exactly. You probably want to have something like this anyways, so that when
someone accidentally unplugs the system, or the disks/CPU/RAM crash, you're not
stuck.
That is, until FreeSWITCH can record its internal state to some inter-machine
memory so we can have hot failover. ;)
-Michael
Fro
or you can put a sip proxy in front of 2 boxes where you can control the
flow of traffic.
when you want to upgrade one, take all the traffic off of it by forcing all
calls to the other box, upgrade it then shift the traffic to the new one.
if that goes well, upgrade the other one too.
On Thu, Ju
Although a little overkill, RFC3398 also describes some desirable interop
behavior between ISUP, ISDN and SIP.
(From "7.2.4.1 ISDN Cause Code to Status Code Mapping")
[...]
ISUP Cause valueSIP response
1 unall
That's what I tried to say, I didn't expressed myself well, sorry.
On Thu, Jun 11, 2009 at 12:09 PM, Steve Underwood wrote:
> Anthony Minessale wrote:
> > One important thing is that if we go around following everything
> > everybody else says
> > we become a follower in our field.
> >
> > I have
Excellent! Thank you everybody. Response 486 did the trick.
Klaus.
Original-Nachricht
> Datum: Thu, 11 Jun 2009 11:31:13 -0500
> Von: Ken Rice
> An: freeswitch-users@lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Rejecting calls without answering
> Hah they are just ret
Ok,
let's try them at half size.
On Thu, Jun 11, 2009 at 11:09 AM, Steve Underwood wrote:
> Anthony Minessale wrote:
> > One important thing is that if we go around following everything
> > everybody else says
> > we become a follower in our field.
> >
> > I have had numerous people tell me what
Hah they are just retrying the call to see if they get a different answer
the 2nd and 3rd time around... This is common unfortunately since 503 in the
VoIP world is typically interpreted by the PSTN world as a "Temp congestion"
(and rightly so) so they will retry and not fail the call... You can tr
Hi Folks,
Here is what i'm observing. When i connect with Xlite (registered device) and
call the 9444 extension (see below), Freeswitch does hangup as i would like it
to.
But when i call via gafachi, something weird happens. What i can see is that
Freeswitch sends a hangup signal (service temp
Anthony Minessale wrote:
> One important thing is that if we go around following everything
> everybody else says
> we become a follower in our field.
>
> I have had numerous people tell me what to do in the code, what to
> name things, what to eat for breakfast.
> Plain and simple, I will choose
It was. You can see the set at line 2 was done before the bridge at line 4.
What am I missing?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 8:06 AM
To: freeswitch-users@lists.free
One important thing is that if we go around following everything everybody
else says
we become a follower in our field.
I have had numerous people tell me what to do in the code, what to name
things, what to eat for breakfast.
Plain and simple, I will choose what to put on our website, when to put
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@lists
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@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log
Are
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.
>
-Ray
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswit
Are you doing an originate or a bridge?
/b
On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote:
In a dialplan, the action sets effective_caller_id_number to a
value, however, in INFO, the displayed value is not the same as the
set. Why?
http://pastebin.freeswitch.org/9361
Thanks, Lars
_
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
-- Meet us at ClueCon! http://www.cl
Hi Team,
I'm still in need of a way to reject a call without answering it. I very much
appreciate your help.
Klaus.
--
GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss
für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02
In a dialplan, the action sets effective_caller_id_number to a value,
however, in INFO, the displayed value is not the same as the set. Why?
http://pastebin.freeswitch.org/9361
Thanks, Lars
___
Freeswitch-users mailing list
Freeswitch-users@li
You can't... once you execute fifo your script has stopped. I think
you have the idea that your script will keep running after you enter
the fifo...
/b
On Jun 11, 2009, at 7:13 AM, Baskar wrote:
>
> Can any one assist me to resolve the above problem
___
Hi,
I have configured inbound in FS SVN Trunk. i have written small program
for inbound call to bridge. i have used session fifo
session.execute( "fifo", "sales_fifo_1 out wait undef
'/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" );
session.execute("bridge", "sofia/inter
Am 11.06.2009 um 05:04 schrieb John Dalgliesh:
Hi,
I am slowly gaining confidence using FreeSWITCH in production, but
there
is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current
calls?
So far I have bee
69 matches
Mail list logo
