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
Andy
Hi,
Is there any reason why the crash-protection parameter in switch.conf.xml
defaults to false and are there any downsides to setting it to true? The
documentation says it helps with certain types of crashes, can anyone tell
me what sort of crashes in particular it helps to prevent as my
Hi,
I have mod_shout installed and I'm using session.recordFile to capture the
audio in a call. When I specify a local file mp3 or wav the audio is
captured fine. However, I'm using an icecast server to manage the audio for
me and when I specify a remote mp3
?
/b
On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote:
Hi Brian,
Just doing some more testing, simplified the call by not even trying to
record the incoming audio and placing a while (session.ready()) {} loop in
the ivr code instead and the calls all now terminate with
RECOVERY_ON_TIMER_EXPIRE
Subject: Re: [Freeswitch-users] Calls being cut off while recording a
message
Please try SVN trunk.
/b
On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote:
Hi Brian,
1.03
Thanks
Andy
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com
] Calls being cut off while recording a
message
Please update... rebootstrap.. you caught SVN with the libtool patch which
kinda broken a few things linking.
/b
On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote:
Hi Brian,
I've upgraded to svn trunk but am now getting errors on load which
recording a
message
Please update... rebootstrap.. you caught SVN with the libtool patch which
kinda broken a few things linking.
/b
On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote:
Hi Brian,
I've upgraded to svn trunk but am now getting errors on load which are
preventing it from working:
2009
network topo?
/b
On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote:
Hi Brian,
Just doing some more testing, simplified the call by not even trying to
record the incoming audio and placing a while (session.ready()) {} loop in
the ivr code instead and the calls all now terminate
Subject: Re: [Freeswitch-users] Calls being cut off while recording a
message
Please try SVN trunk.
/b
On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote:
Hi Brian,
1.03
Thanks
Andy
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com
Hi,
I'm using freeswitch as a glorified answering machine. FS registers with a
VOIP gateway and all calls into the gateway go through an ivr menu and are
allowed to leave a message which gets recorded to a file. The FS box is
behind a NAT firewall. Everything works fine except that
being cut off while recording a
message
I'm going to guess you're not on SVN trunk? what rev are you on?
/b
On Mar 31, 2009, at 8:04 AM, Andy Ayers wrote:
Hi,
I'm using freeswitch as a glorified answering machine. FS registers with a
VOIP gateway and all calls into the gateway go through
, at 10:07 AM, Brian West wrote:
Upgrade to 1.03 or SVN Trunk
/b
On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote:
Hi,
I've recently ugrade to version 1.02 of freeswitch and am having some
problems with my gateway registrations. The gateway successfully registers
with my voip provider when
Gateway registration
Upgrade to 1.03 or SVN Trunk
/b
On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote:
Hi,
I've recently ugrade to version 1.02 of freeswitch and am having some
problems with my gateway registrations. The gateway successfully registers
with my voip provider when freeswitch first
Hi,
I've recently ugrade to version 1.02 of freeswitch and am having some
problems with my gateway registrations. The gateway successfully registers
with my voip provider when freeswitch first starts but if left running it
seems to loose it's connection to my voip provider. I can get it to
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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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,
I'm using freeswitch to receive incoming calls from a sip provider namely
AQL. When my freeswitch box is connected directly to the internet everything
works fine. When I place a firewall/router inbetween the box and the
internet, the software registers with the sip provider ok and answers
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 any good VOIP providers
Hi,
Can anyone recommend any good VOIP providers that integrate well with
Freeswitch. In particular I need one that can cope with Freeswitch being
behind a firewall/router. I've tried Voiptalk.org and voipon.co.uk but
neither seem to register correctly via the router.
Any help much
Hi,
This is my first post so my apologies if I get the protocol wrong or if I'm
posting to the wrong place.
Has anyone any experience of setting up Freeswitch to accept incoming calls
via VoipTalk (http://www.voiptalk.org) through a firewall/router.
I have tried every possible combination of
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