When I uncomment the following tag, internally held calls no longer hear
MOH.
param name=inbound-bypass-media value=true/
Is there a way to have the above uncommented and still provide MOH to held
calls?
Best Regards,
Jerry
___
FreeSWITCH-users
in the sangoma driver that has been fixed it the latest
release.
On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards jerry.richa...@teotech.com
wrote:
Hello All,
I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644.
I am still having the problem where a PSTN-to-Internal call via
Okay. I uncommented the following lines and the video start works as
correctly:
param name=media-option value=bypass-media-after-att-xfer/
param name=inbound-bypass-media value=true/
Thanks,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent
Is there a setting to control how fast FS distributes presence changes to
subscribers? Currently, it appears to take several minutes before I see
presence changes. I would like to see them almost instantaneously, if
possible.
Thanks and Best Regards,
Jerry
I have a Freeswitch PBX server with an installed Sangoma A101D card
connected to a PRI. Most everything works okay, however when I get an
inbound call from the PSTN, if the call is not answered within about 12
seconds, the call ends (so it doesn't go to voice mail). If I make a call
from one
Jerry Richards schrieb:
After establishing an audio call between two Bria softphones, and then
starting video at the caller phone, FS replies to the re-INVITE with a
200 OK with only the PCMU codec. This looks incorrect. The audio
call previously negotiated to the speex/16000 codec
Is it possible to allow/deny REGISTER requests based on the User-Agent
header? I need to know/manage what devices are registering.
Best Regards,
Jerry
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sofia profile siptrace on replace
with your profile.
/b
On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote:
I upgraded to the latest 1.0.5pre9 and now if I try to call from an
internal phone to an external number on my Sangoma PRI, I get a 502 Bad
Gateway
reply. Below is the console
I upgraded to the latest 1.0.5pre9 and now if I try to call from an internal
phone to an external number on my Sangoma PRI, I get a 502 Bad Gateway
reply. Below is the console loglevel 7 output. It says the destination is
out-of-order. I'm not sure what this means. Any help is appreciated.
I am trying to bring up a video call, but not having much luck. We are only
getting one-way video (i.e. the caller sees far-end video, but the callee
does not). I added the H263/H264 tags to the pre-process
global_codec_prefs and outbound_codec_prefs tags in vars.xml.
Anyone have hints on
Anthony and Michael,
I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs
that Anthony told me to turn on. I put the results up in the PasteBin.
Best Regards,
Jerry
_
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, December 07, 2009
Here is the Pastebin Link: http://pastebin.freeswitch.org/11432
Thanks,
Jerry
_
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Tuesday, December 08, 2009 12:35 PM
To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Machine
[mailto:m...@jerris.com]
Sent: Saturday, December 05, 2009 7:30 PM
To: Jerry Richards
Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
UNREACHABLE When Gateway Sends RTP
Jerry-
Any update on this?
Mike
On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote:
Why are you changing
/freeswitch/trunk freeswitch
Best Regards,
Jerry
_
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, December 07, 2009 7:44 AM
To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
UNREACHABLE When
I have Mediant 1000 gateway, and for some reason, when I make an outbound
call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A
Wireshark trace shows that FS is replying to the gateway's inbound RTP
packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP
Hello,
I just pasted a log in the Pastebin with Freeswitch logging enabled. Does
anyone know a way to prevent FS from connecting the call prior to the callee
answering?
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Thursday
Subject: Re: [Freeswitch-users] Accessing Config Info From Database
On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
I have a bit of confusion about Lua scripting. When a script is invoked,
should it always return an XML string that is used by FS
/Mod_xml_curl
or mod_xml_odbc (generate xml in freeswitch):
http://wiki.freeswitch.org/wiki/Mod_xml_odbc
or LUA together with luasql (generate xml in freeswitch):
http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration
regards,
Leon
On Fri, 2009-11-13 at 13:59 -0800, Jerry Richards wrote
How can I disable MD5 Authentication upon registration?
Best Regards,
Jerry
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Is there a way to access configuration information from a database (e.g.
SQL) rather than from the XML files?
Best Regards,
Jerry
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FreeSWITCH-users@lists.freeswitch.org
Here is what is believed to be a bug found by Robert Hadley found in
Freeswitch1.0.4/scripts/gentls_cert.in build file:
Fix for gentls_cert remove to work:
[scripts]# diff gentls_cert.in gentls_cert.in~
129c129
if [ -d ${CONFDIR}/CA ]; then
---
if [ ! -d ${CONFDIR}/CA ];
I am trying to make a call through a Gateway that sends the INVITE with no
SDP and ONLY wants the 200 OK w/SDP when the callee answers.
For some reason, Freeswitch answers the call with 200 OK w/SDP even before
the callee answers the phone. Is this to provide ringback? Can I disable
that
as it is
going to get. For delivering a service you'd probably want interaction with
a DB. I've use XML curl a lot and have even starting using direct DB
queries from static dialplans using mod_memcache and memcachedb (not
memcache ... persistent storage).
SDR
Jerry Richards wrote:
My understanding
When I start Freeswitch, I see an Error checking for PMP [general error]
as shown below. Does anyone know what could cause this?
[r...@teoproxy bin]# ./freeswitch
Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch
-waste.
auto-adjusting stack size for optimal
My understanding of DialPlan/CallRouting is that it can be accomplished via
static XML tags, or alternatively, via a DialPlan Application that
interfaces with the dptools module.
Question: If my above assumption is true, how does one select one approach
over the other? What is the
I have my Freeswitch server with an installed Sangoma A101D card. Most
everything works okay, however, when I get an inbound call from the PSTN, I
see the following warning show up in the log. Additionally, the caller (on
the PSTN) does not hear ringback, and if the call is not answered within
I notice that when I call IVR from the PSTN, the Welcome to Freeswitch...
introduction is clipped at the beginning, so it sounds like come to
Freeswitch. If I call 5000 internally, then I always hear the full
introduction. What can I do to resolve this?
My XML config looks like:
extension
1000 ms... we usually bring up media too fast before the other end is
ready.
/b
On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
I notice that when I call IVR from the PSTN, the Welcome to
Freeswitch...
introduction is clipped at the beginning, so it sounds like come to
Freeswitch
Did the voting booth close? I was unable to vote. I'm not sure what link
to click and I have had some strange issues with my FS account today.
I would be interested in paid training. Do you have plans for offering a
training session at your locale? Or would you travel onsite to provide
I installed FS on a machine with a Sangoma A101D (PRI) card and if I make an
inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN
phone. If I call IVR from an internal phone, then it does recognize the
DTMF digits. I have mostly default configurations for everything.
Sometimes if I stop (using ... command) and then restart freeswitch (using
./freeswitch command), the program will crash and return to the Linux
(CentOS 5.3) prompt. I am using version 1.0.4.
I just pasted the freeswitch/terminal log into the Pastebin.
Best Regards,
Jerry
Can anyone recommend a good 3rd party dialplan tool that will work with
Freeswitch?
Best Regards,
Jerry
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
How do I use the dial plan to add leading digits to an outgoing call through
a gateway?
My internal phone number is 5380, but when FS sends the call to the gateway
I want the CALLING party to be 4253495380.
Best Regards,
Jerry
___
FreeSWITCH-users
-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, October 19, 2009 9:14 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: Adding Leading Digits To CALLING Number of Outgoing Call
How do I use the dial plan to add leading digits to an outgoing call through
I am building Freeswitch on a Centos 5.3 machine and the last step below
gets an error because there is no scripts/contrib folder. Anyone know why?
./configure
make
make all install sounds-install uhd-moh-install moh-install
scripts/contrib/trixter/makemodconf.pl modules.conf
Okay, I think the contrib folder moved up one level. So the Wiki
installation documentation should probably be updated to reflect that.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, October 16, 2009 9:47 AM
implementation?
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, October 05, 2009 3:24 PM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] SLAs and BLAs
We are building our own in-house developed Teo
. They are the same company that make the Eyebeam.
Any ideas?
Best Regards,
Jerry
_
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, October 02, 2009 11:28 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay
are missing the trailing ;
On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards jerry.richa...@teotech.com
wrote:
I put the sqlite3 select query in the paste bin, and prior to that, I
entered the .dump command. The select command came back with a ...
prompt which I don't understand. I don't know
I can see how BLFs and Presence are managed, however I haven't found much
documentation on SLAs and BLAs. What is the RFC(s) that Freeswitch used to
implement SLAs and BLAs? Do they differ from BLFs?
Best Regards,
Jerry
___
FreeSWITCH-users mailing
:02 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLAs and BLAs
First off what phones are you going to be using?
/b
On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:
I can see how BLFs and Presence are managed, however I haven't found
much documentation on SLAs
How would I configure FS to Call Forward All or Call Forward when Busy or
Call Forward when No-Answer? Can this be done at the server, rather than at
the phone?
Best Regards,
Jerry
___
FreeSWITCH-users mailing list
...@freeswitch.org]
Sent: Thursday, October 01, 2009 8:14 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones
Piece of advice, don't ask, just do it. ;)
jmesquita
On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa
%') and
(sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)
On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards jerry.richa...@teotech.com
wrote:
Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional
...
[WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: FS Does Not Relay
Presence PUBLISH
ToSubscribing Phones
which phone is it,
we tested it with eyebeam and it appears to work for us.
On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
By the way, I see the following lines at the FS console, which might be a
clue as to why
If you have time to take a look, I could put a trace in the pastebin?
Jerry
_
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Thursday, October 01, 2009 10:29 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay Presence
I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone. Does anyone know why?
Best Regards,
Jerry
Sorry for this mundane question, but how do I search mailing archives for
keywords? The following link has no search option?
http://lists.freeswitch.org/pipermail/freeswitch-users/
Thanks And Best Regards,
Jerry
___
FreeSWITCH-users mailing list
Is there a way in FS to selectively deny a BLF presence subscription
request for the sake of privacy? So that groups could be defined that are
allowed to be monitor or be monitored? And others that are not allowed to
monitor or be monitored?
Best Regards,
Jerry
Hello All,
I have an internal extension that needs to send an INVITE without SDP body
(Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable
with reason MANDATORY_IE_MISSING. Would anyone know what I need to do to
enable this?
Best Regards,
Jerry
Hi,
Does Freeswitch support OCS? We are interested in having our desktop PC
control our in-house desktop phones (e.g. initiate call, answer call, hold
call, etc.) using the uaCSTA protocol.
Best Regards,
Jerry
___
FreeSWITCH-users mailing list
ships with presence enabled for SIP
if you have a phone that supports it, all you have to do is enable it on the
phone.
On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards jerry.richa...@teotech.com
wrote:
Also, is presence conveyed as any string? Or is presence a predefined list
of status
I would like to modify my SIP phone and my gateway to convey/exchange
presence information. Could someone point me toward the FS presence
documentation? I've seen bits and pieces. Also, I think presence can be
communicated via more than one protocol.
Thanks And Best Regards,
Jerry
Also, is presence conveyed as any string? Or is presence a predefined list
of status?
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Tuesday, September 15, 2009 8:46 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: FS
What account do I need to create to post logs in the Pastebin? I tried my
mailing list username/password, and also tried a jira.freeswitch.org
username/password. Neither of these were accepted.
Best Regards,
Jerry
___
FreeSWITCH-users mailing list
Aha... I have been notified that I failed the test. The username/password
is given in the authentication pop-up itself. My bad...
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, September 14, 2009 8:13 AM
To: 'freeswitch-users
default=deny
node type=allow cidr=192.168.72.186/32/
...
/list
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 1:27 PM
To: 'freeswitch-users@lists.freeswitch.org'; 'Michael Collins'
Subject: RE: Inbound
I am trying to configure a Grandstream gateway to work with FS. I can make
outbound calls without a problem. However, inbound calls are getting a 403
Forbidden from FS in response to the INVITE from the gateway.
Now, the INVITE's from address is the caller's number (e.g. 111222),
which
By the way, the FS DEBUG console is saying the following when an inbound
call is made:
Rejected by acl domains. Falling back to Digest auth.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 10:25 AM
I have phones registered internally and can call among them. However, when
I dial 711 from an internal phone, freeswitch replies with 484 Address
Incomplete with reason INVALID_NUMBER_FORMAT. At the server console, I
see the following error:
[ERR] mod_sofia.c:2645 Invalid Gateway
Does
. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com
On 05-Sep-2009, at 12:03 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
Under the Minimum/Recommended System Requirements, what is meant by
We recommend you plan for 50% duty cycle? What is this duty
Under the Minimum/Recommended System Requirements, what is meant by We
recommend you plan for 50% duty cycle? What is this duty cycle?
Also, I see that the system requirements indicate Freeswitch recommends 1GB
RAM and 50MB disk space. I guess I'm wondering how the number of extensions
and
I submitted this to the dev-list, but maybe it should be in the user-list:
Can I register two phones to the same Line-ID? That is, does Freeswitch
support a configuration where multiple endpoints have the same extension
number, auth-id and password? And if so, do I have control over whether an
Does Freeswitch support Presence via SIMPLE protocol? Can it maintain
presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement?
Best Regards,
Jerry
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Hello All,
Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
how many users?
Also, when that number is exceeded, how can Freeswitch server be distributed
to accommodate a larger installation?
Best Regards,
Jerry
___
Hello All,
I noticed Freeswitch becomes the middle-man, handling RTP traffic for an
active call. How do I configure it so it allows the two SIP endpoints to
send RTP packet to each other directly?
Best Regards,
Jerry
___
FreeSWITCH-users mailing
Hello All,
I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine
for the first time using the Getting Started Guide. I can register three
lines (1000, 1001, and 1002), but when I attempt to call one phone to the
other I hear the operator say:
The person at extension 1000
: [FACILITY_NOT_SUBSCRIBED]
Are you trying to test everything on the same machine?
/b
On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:
Hello All,
I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP
machine for the first time using the Getting Started Guide. I can
register
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