Thanks Brian, will check it out.
I am using FS as Voicemail behind Opensips. As we have 2 Opensips
servers if FS responds with a Contact header with a URI value we
cannot route the call back to the correct FS server and the call is
eventually dropped. For some reason this occurs even though we
I could not get this working on current trunk. Can you post your
configuration on conference module and the dialplan example?
Thanks,
jmesquita
On Thu, May 28, 2009 at 12:56 PM, Michael Collins m...@freeswitch.orgwrote:
On Wed, May 27, 2009 at 8:00 PM, j3flight jcro...@gmail.com wrote:
Sorry for my presentation.
Call - is nothing but the outbound number in the far end. (this is the leg I
am trying to transfer)
Extension - is nothing but the internal softphones
My internal extension are not hanging up.
On Thu, May 28, 2009 at 8:51 PM, Anthony Minessale
Ok Thanks Mike.
I hope in asterisk it is not there. I was trying some couple of things,
1. when I use Java servlet to dial a call thru asterisk using manager
interface it slows down.
2. when I use the same Java servlet with Freeswitch I feel the speed and
usage in the tomcat. Its bit faster for
After ZRTP negotiation is complete (the ZRTP state machine has entered the
secure state), I get a number of lines in the log as follows (FreeSWITCH
rev. 13501):
2009-05-29 16:43:19 [DEBUG] switch_rtp.c:538 zrtp_logger() [zrtp protoco]:
ERROR! Decrypt failed. ID=14:DH s=SRTP authentication
After using the latest trunk revisions I get no audio anymore. The last working
build I have is about 5 days ago. I havn't upgraded until today, so I don't
know exactly when this happened.
I've noticed quite a few changes on the RTP stack, beacuse of the
implementation om ZRTP, and I guess
Sorry for missing this in my last post, but I'm using sofia for all calls.
/Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Peter Olsson
Skickat: den 29 maj 2009 12:31
Till: 'freeswitch-users@lists.freeswitch.org'
Ämne:
Peter Olsson peter.ols...@visionutveckling.se wrote:
After using the latest trunk revisions I get no audio anymore. The last
working build I have is about 5 days ago. I havn't upgraded until today, so
I don't know exactly when this happened.
You could always check out some intermediate
Hi, my job with FS has been on Windows.
Diego
--- On Thu, 5/28/09, Brian West br...@freeswitch.org wrote:
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] The calls are dropped during register
To: freeswitch-users@lists.freeswitch.org
Date: Thursday, May 28, 2009, 8:22 PM
And mine with the same behaviour on Linux.
Best regards
Peter
Diego Toro schrieb:
Hi, my job with FS has been on Windows.
Diego
--- On *Thu, 5/28/09, Brian West /br...@freeswitch.org/* wrote:
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] The calls are
try enabling the Path header too
we fully support that
On Fri, May 29, 2009 at 1:37 AM, Jim Burke j...@evolutiontel.net wrote:
Thanks Brian, will check it out.
I am using FS as Voicemail behind Opensips. As we have 2 Opensips
servers if FS responds with a Contact header with a URI value we
There is a very long explanation as to the differences behind asterisk AMI
and FreeSWITCH Event Socket that I will not get into now
but it's not related to APR or TCP socket performance whatsoever it's more
about asynchronous versus monolithic modeling.
On Fri, May 29, 2009 at 4:42 AM,
maybe you have a nat issue sending byes to the phones
enable debug log by pressing f8 or typing console loglevel debug
and turn on sofia trace with sofia profile internal siptrace on
capture the entire thing and paste it to
http://pastebin.freeswitch.org
On Fri, May 29, 2009 at 4:37 AM,
Its called build skew... we added an extra_data element to the frame
struct. Please do a fresh checkout and build.
/b
On May 29, 2009, at 5:46 AM, Jason White wrote:
Peter Olsson peter.ols...@visionutveckling.se wrote:
After using the latest trunk revisions I get no audio anymore. The
This is normal because the switch from clear to secure can happen
quickly on one end or the other and you'll have a few packets that get
thru before one end is ready... nothing to be worried about.
/b
On May 29, 2009, at 5:02 AM, Jason White wrote:
After ZRTP negotiation is complete (the
Peter find me on IRC and let me into your machine so I can trouble
shoot this.
Thanks,
Brian
On May 29, 2009, at 7:58 AM, Peter P GMX wrote:
And mine with the same behaviour on Linux.
Best regards
Peter
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
I would recommend not turning on the SLA option then... I had to add
that in because when using TLS the phone would try to call the IP
which would fail because the SSL cert wouldn't match and the poor
phone would kill over, lock up and reboot sometimes :P GO POLYCOM!
With that option not
I've looked into this a bit more now, and I think it is a sofia issue, I will
look trough the changes in sofia since I had the last working configuration,
and see if I find anything.
/Peter
-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org
Nope its not a sofia issue... its build skew ;)
/b
On May 29, 2009, at 8:24 AM, Peter Olsson wrote:
I've looked into this a bit more now, and I think it is a sofia
issue, I will look trough the changes in sofia since I had the last
working configuration, and see if I find anything.
I did that, and it compiles fine. It's just not working :) But as I said in my
last post, I think it could also be related to sofia, when using h323 it
works... However - maybe I'm using opal's RTP stream by then..?
I'll get some logs for the scenario, and if I don't find a solution I'll start
Nope - it's not :)
Just to make sure I even deleted the source completely, and checked everything
out again.
/Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 29 maj 2009 15:26
Till:
when you say i did that
you typed make current to rebuild?
or you are assuming your successful compile is the same effect as cleaning
the 100 object files
that have the wrong data structure in them so the audio data they really
seek is 8 bytes offset from where they think they are
until they are
did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too
?
On Fri, May 29, 2009 at 8:33 AM, Peter Olsson
peter.ols...@visionutveckling.se wrote:
Nope – it’s not :)
Just to make sure I even deleted the source completely, and checked
everything out again.
/Peter
Actually I deleted everything from disk and downloaded a fresh clean copy from
SVN and rebuilt it from scratch. I should mention that I'm on windows, so I
never do make current. I just do a full clean, get latest from SVN and
rebuild, that's what I do every time. But for this time I even
I'm on Windows, so I have everything under my fs directory, but I deleted the
complete directory and did everything from scratch...
/Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009
Hi Guys,
I have install WANPIPE Release: 3.5.2 for sangoma A104D and FreeSwitch
1.0.4pre8 with openzap modul.
and I use Lua script for playing wav file.
and I got error like this below while I call the number 0312982300, if i
run ./fs_cli , the FS can pickup a call for moment, after more than 1
Hi Anthony,
I updated to rev 13496 -- now I have a different problem... I connect to
the event socket interface, ask for all events... then never receive any
events!
From telnet:
Content-Type: auth/request
auth ClueCon
Content-Type: command/reply
Reply-Text: +OK accepted
events plain all
Can you try to do a binary search and nail down the exact version that
caused this issue and then file a bug on http://jira.freeswitch.org.
Thanks
Mike
On May 29, 2009, at 9:55 AM, Peter Olsson wrote:
I’m on Windows, so I have everything under my fs directory, but I
deleted the complete
On Thu, May 28, 2009 at 10:37 PM, Marc Orenberg m...@kasteris.com wrote:
Hi, I'd like FreeSWITCH to be able to communicate with a Musicam Prima LT
device. (http://www.musicamusa.com/products/prima/PrimaLT.htm). This is a
POTS codec, which (I've just learned) means that the connection is made
I'd be very interested to see your script and dialplan , for me it's a
very important issue as my conference server facing to outside and I
need to have moderators and regular users entering different pins.
Anton
jcro...@gmail.com wrote:
Unfortunately, the instance of FreeSwitch where I've
Maybe someone can do a wiki page with scripts and a howto?
/b
On May 29, 2009, at 3:31 PM, Anton Karpov wrote:
I'd be very interested to see your script and dialplan , for me it's a
very important issue as my conference server facing to outside and I
need to have moderators and regular users
I'll post the script, dialplan and how-to on the wiki as soon as I
can...
Will follow up on this thread with a link once it's complete.
Thanks for the interest.
Brian West wrote:
Maybe someone can do a wiki page with scripts and a howto?
/b
On May 29, 2009, at 3:31 PM,
Hi:
I get this error when voicemail try to send an email:
'/bin/cat /tmp/mail.12436473394319 | sendmail -t (null)'
This is the called extension:
include
user id=1010 mailbox=1010
params
param name=password value=$${default_password}/
param name=vm-password value=1010/
Are you really using sendmail or are you using something like exim?
/b
On May 29, 2009, at 8:58 PM, Luis M. Zuccolo wrote:
Why the vm_mailto variable isnt't passed to the script?
What's wrong?
Someone can assist me?
Thanks in advance
Luis Zuccolo
Brian West
br...@freeswitch.org
-- Meet
Brian West br...@freeswitch.org wrote:
This is normal because the switch from clear to secure can happen
quickly on one end or the other and you'll have a few packets that get
thru before one end is ready... nothing to be worried about.
I thought that might be the scenario.
In a typical
If you happen to have a polycom or snom and you use the new sched_api
extension I added to trunk (commented out) it will sched_api and snag
the zrtp sas1 and sas2 strings and 4 seconds after the call is up
update the display of the polycom with those two strings... kinda
handy eh?
For
I'm using postfix, that has a compatiblilty interface to sendmail.
On Fri, 2009-05-29 at 21:36 -0500, Brian West wrote:
Are you really using sendmail or are you using something like exim?
/b
On May 29, 2009, at 8:58 PM, Luis M. Zuccolo wrote:
Why the vm_mailto variable isnt't passed
Luis M. Zuccolo luismzucc...@yahoo.com.ar wrote:
I'm using postfix, that has a compatiblilty interface to sendmail.
I've used this with Sendmail successfully; it should work with Postfix too.
See the mailer-ap and mailer-app-args variables in
autoload_configs/switch.conf.xml and be sure they
Jason White ja...@jasonjgw.net wrote:
See the mailer-ap and mailer-app-args variables in
autoload_configs/switch.conf.xml and be sure they are set correctly for your
installation. Try running the Postfix sendmail program manually to be sure
that it is working correctly.
sendmail -t is the
Yes, in console works well (without null).
This variables are sets in switch.conf.xml:
param name=mailer-app value=sendmail/
param name=mailer-app-args value=-t/
Previously I've omitted the error:
sh: -c: line 0: syntax error near unexpected token `('
sh: -c: line 0: `/bin/cat
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