I'm using VQManager (there is a 30 day trial) and it's useful for seeing who
does what / when per call; it's very easy to install...
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Frank @
Impact
Sent: Thursday, December
Skype have opened their beta program up to all comers.
http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent
Three lines in a sip_profile make FreeSWITCH talk nicely; but using the
PCMU codec.
Any progress on SILK native support? Last I saw was discussion back
- and really just exposing a problem I've always had
before?
The config is (50 Polycom Phones - NAT - Internet - Amazon EC2)
I would really appreciate some pointers on what to look for; additional
trace that might reveal something.
Thanks, Chris.
On Fri, 16 Oct 2009 20:06:52 -0700, Chris Fowler ch
Hi,
We've been using 13168M in production for some time now (works great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE]
Hi Ray,
This was a problem some time ago (couple of months ago). Are you running the
latest build?
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Raymond
Chandler
Sent: Wednesday,
I'm using FreeSWITCH (Build 13168M) and we're having intermittent multi-second
delays on conference bridges with more than three participants (this is not a
new issue - just bubbled to the top of the stack to address).
The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 with
, 2009 08:29
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Using Variables in Dialplans
On Wed, Apr 22, 2009 at 5:20 PM, Chris Fowler ch...@fowler.cc wrote:
Brian, Michael,
Thanks for the help - I had read that but not fully comprehended it until you
spun it the way
I have the following defined:
!-- Billing Open? --
extension name=billing_open continue=true
!-- man strftime - M-F, 9AM to 5PM --
condition field=${strftime(%w)} expression=^([1-5])$/
condition field=${strftime(%H%M)}
expression=^((09|1[0-6])[0-5][0-9]|1700)$
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, April 22, 2009 13:10
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Using Variables in Dialplans
On Wed, Apr 22, 2009 at 12:19 PM, Chris Fowler ch...@fowler.cc wrote:
I
] switch_core_session.c:1020
switch_core_session_thre
ad() Close Channel sofia/internal/1...@**.***.219.221 [CS_DONE]
Thank you for the help.
~Alex
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
About two weeks ago FlowRoute stopped working with FreeSwitch. Looking at the
SIP trace there is chatter about a Proxy Auth required. I ran out of time to
debug as this was on a production system.
The work around was:
1) You need to prefix the dialed number with the FlowRoute Account Number
Are you either restarting FS or issuing the reloadxml command (press F6 on the
console) after making these changes?
Did you modify vars.xml per my last note?
Also, check out http://wiki.freeswitch.org/wiki/Getting_Started_Guide - it's
worth investing the time to understand how FS parses the
I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom
phones working happily in this config.
I modified the Internal profile in
/usr/local/freeswitch/conf/sip_profiles/internal.xml to include:
param name=aggressive-nat-detection value=true/
param name=NDLB-force-rport
Brian: Did you request public IP's for your EC2 instance?
Yes; there is an Elastic IP (EIP) associated with the instance.
Also specify the EIP in vars.xml
X-PRE-PROCESS cmd=set data=bind_server_ip=insert EIP here/
X-PRE-PROCESS cmd=set data=external_rtp_ip=insert EIP here/
Sent: Wednesday, March 25, 2009 12:43
btw you'll have to reinstall your phrase macros make vm-sync I
think should do it if it doesn't let me know... we removed the 250ms
sleeps and that was the problem which we fixed.
I re-did the macros; the only change I could detect was the
Did you provide the menu you are using and what you expect to happen?
Here's the setup;
Caller - FlowRoute - FreeSwitch
menu name=main_ivr
greet-long=phrase:welcome
greet-short=phrase:top-menu
invalid-sound=ivr/ivr-that_was_an_invalid_entry.wav
Any thoughts on why FS saw all digits 1029 but only reports '029'?
2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect()
digits '029'
Config:
menu name=main_ivr
greet-long=phrase:welcome
greet-short=phrase:top-menu
First off what SVN rev? Remember when reporting issues try to include all
the information you can!
Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647)
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Thanks for the tip Brian.
Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log
Chris.
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Hi,
Ive been seeing an issue where FreeSWITCHs CPU and memory utilization
climb over time; a restart of FS clears up the problem.
See graphs for the past week. http://cfowl.postinbox.com/fs.jpg
Observed on the Release Candidate, and then upgraded to the current
trunk a couple of times.
Jay : what happens in your dialplan ?
Nothing special; no external script execution just default pattern
matching to route to extensions (per the stock config).
Brian: Can you update to SVN trunk as of now?
Yup, I will pull the trunk and report back in 24 hours.
Chris.
Brian: Can you update to SVN trunk as of now?
I updated - version reports: FreeSWITCH Version 1.0.trunk (12631)
Only difference I note with this build is that upon shutdown FS now
SegFaults. The mem/cpu usage continues to slowly climb.
snip
2009-03-16 20:59:32 [CONSOLE]
22 matches
Mail list logo
