Just want to say that you were right. Updating to trunk solved the problem.
It seems that I updated/rebuilt my copy just before the patch was
applied on 4 September.
Thank you again!
On Thu, Sep 10, 2009 at 9:54 AM, Vassil Panayotov
panayotov...@gmail.com wrote:
Michael, Moises and Octavio
meftah,
i disabled mod_erlang_event in modules.conf. unixodbc is installed already.
still ... the same error message. tks for your input.
/nandy
On Sun, Sep 13, 2009 at 11:56 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:
hello,
i think you enabled mod_erlang_event in the modules.conf
Hi,
I'm newbie in FS. I want to know how to Filter the string to include only
the allowed characters in FS?
Kindly advice me.
--
Regards,
Ahmed Munir
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Hi,
Does anyone have any luck on porting freeswitch to blackfin + uclinux?
Is this a feasible option?
jb
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You have a merge conflict please svn revert sofia.c
/b
On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote:
Hi Folks,
I've got a compilation error with the latest codes (r14842)
Making all in packages
Creating mod_sofia_la-mod_sofia.lo
Compiling mod_sofia.c ...
Creating
Hi
I need to measure DTM digits duration and interdigit delay for various
phones in a two stage dialing scenario. I.e Phone dials DID and after answer
then the second number
My set-up is:
Phone-PSTN network-DID(inband DTMF) -FS
I ha ve FS to answer the call and record the call - all this is
_
My suspicion is that this is only for zaptel type cards. Our tests
with Sangoma analog cards have all been pretty successful. But thanks
for info! Anyone else using Rhino, Digium, or compatible analog cards?I am
not experiencing an audio delay. My configuration is exactly as
Hi,
A client of ours is trying to connect his * to our FS, outgoing calls
work fine, unfortunately when we try to forward an incoming call to his
* it's not going through.
I see his registration in our internal profile which looks just fine.
We try to forward incoming calls using this in FS
Hi,
I tried to performance test freeswitch with media proxy thur fs. With 400
cps, I start to see 2000 channels remaining in Freeswitch, and then no read
codec error starts to pop up. With only 1875 channels, how come freeswitch
is complaining about no read codec? Also, I am using media_proxy
Swapping hardware... I've noticed other odd things... Things that
shouldn't happen, do.. But not consistently The phrase,
It's computing Jim, but not as we know it... pretty much describes
the situation.
Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3231 x0
On Sep
The first hint was when the firmware rev began with the letters POS
On Mon, Sep 14, 2009 at 8:15 AM, Karl Vesterling k...@ken-ton.com wrote:
Swapping hardware... I've noticed other odd things... Things that
shouldn't happen, do.. But not consistently The phrase, It's
computing
After I paused the traffic from sipp and when sipp finished, I still got
a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the
cause of these zombies and how can I fix it?
One way might be to not DDoS your box at 400cps?
(You are out of rtp ports *and* you are pushing your
This means the far end is sending you a challenge and we do not know
how to answer it... please review how to setup a gateway on the Wiki
so you can authenticate.
/b
On Sep 13, 2009, at 6:47 PM, paul.d...@gmail.com wrote:
Only abnormal things I can see in FS logs are:
2009-09-13
HAHA I couldn't have said this better!
/b
On Sep 14, 2009, at 8:17 AM, Anthony Minessale wrote:
The first hint was when the firmware rev began with the letters POS
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On Sep 13, 2009, at 9:27 AM, Morten Henckel wrote:
However when i analyse the rdecording the Digits are being cut off
down to 10 msec bursts - I trust its FS that cust the DTMF in
order to avoid further propogation inband to second leg of the call.
Nope if its rfc2833 its not us cutting
Hi Woody,
well, it is quite hard to answer you back with this logs...
you didn't tell us:
1. what machine are you running (CPU/RAM)
2. what distro are you running - 32 or 64 bit (i had some lets say
experience with a wrong selection :P)
3. what is your configuration
Hi,
i just have a maybe dummy question but it is still a question :P
*action application=record
data=${recordpath}/${service_instance}/${record_filename} 20 200/*
in my case ${service_instance} is something dynamic and has to be created on
the fly.
Is there any way FS can create a
Ahmed,
if you are talking about dial patterns then yes, freeswitch takes you a
mile ahead and utilizes regular expressions for pattern matching, you
could probably use something like this: ^([0-9]+)$
above simple regex will allow any digit from 0 to 9 and + indicates
repetitive, so this
Thank you for the hint.
But.. why would I need a gateway in this case? I am just trying to ring an FS
extension, right?
Anybody has a clue how to make * not to send the challenge?
This means the far end is sending you a challenge and we do not know
how to answer it... please review how to
yep,
just sow it in the meantime... thanks.
btw: can i use mod_shout to stream files to a server..
e.g. *action application=record data=shout://
server.domain.com/${recordpath}/${service_instance}/${record_filename} 20
200/*
can it work?
T.
On Mon, Sep 14, 2009 at 4:15 PM, Leon de Rooij
it's a rite of passage :)
On Mon, Sep 14, 2009 at 11:29 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
Aha... I have been notified that I failed the test. The username/password
is given in the authentication pop-up itself. My bad...
-Original Message-
From: Jerry
Hello FreeSWITCHers!
We are looking for people who are in a position to help out with various
subprojects that will help FreeSWITCH to keep growing. We need people to
help out in these basic areas:
Bug marshals (people who watch JIRA and test bug reports, patches, etc.)
Documentation maintainers
What account do I need to create to post logs in the Pastebin? I tried my
mailing list username/password, and also tried a jira.freeswitch.org
username/password. Neither of these were accepted.
Best Regards,
Jerry
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Aha... I have been notified that I failed the test. The username/password
is given in the authentication pop-up itself. My bad...
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, September 14, 2009 8:13 AM
To:
This works for me:
action application=mkdir data=${filebase_dir}/
You must set ' filebase_dir ' before.
- Original Message -
From: Tihomir Culjaga
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, September 14, 2009 4:58 PM
Subject: [Freeswitch-users] FS create
nice ... thx.
T.
On Mon, Sep 14, 2009 at 4:41 PM, Evgeniy Zolotov zolo...@altron.ua wrote:
This works for me:
action application=mkdir data=${filebase_dir}/
You must set ' filebase_dir ' before.
- Original Message -
*From:* Tihomir Culjaga tculj...@gmail.com
*To:*
Try username pastebin with pasword freeswitch (without )
Jerry Richards wrote:
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,
There is no such things as FS 1.4, but 1.0.4 yes.
On Mon, Sep 14, 2009 at 2:20 PM, paul.d...@gmail.com wrote:
Thank you for the hint.
But.. why would I need a gateway in this case? I am just trying to ring an
FS extension, right?
Anybody has a clue how to make * not to send the challenge?
Hi,
I am using the function session.collectInput and session.streamFile to
collect a number of DTMF digits.
If the DTMF digits are sent in the RTP, i can collect several digits until
timeout. No problem there! If the DTMFs are received as a sequence of SIP
INFO packages, collectInput only
We currently don't support forked dialogs.
Mike
On Sep 8, 2009, at 12:16 PM, Humberto Quintana wrote:
Hi Brian,
Thank you very much for your answer but both, Freeswitch and
Kamailio have public IPs, it's my NAT'd IP phone who has private IP
but this is fixed by Kamailio.
The problem is
Okay. I got the Grandstream Gateway's 1-stage dialing working with
Freeswitch (Thank You, Michael Collins and Thank All You Developers for
creating this really slick Softswitch/PBX).
Here are the changes/additions I made to the XML files:
conf/sip_profiles/exernal/grandstreamGXW4104.xml (added
You can assign two things to me.
1. libesl code documentation (partially done and Doxygened - needs cleaning)
2. Bug marshal. I am setting up the proper lab environment here to be able
to test most stuff.
Count me in for any questions I can answer and I am _always_ on IRC
jmesquita
On Mon, Sep
I just committed revision 14849 to make sip_invite_params only apply
to the RURI, If you wish to modify the To param son the invite you
MUST use sip_invite_to_params moving forward.
you have sip_invite_contact_params and sip_invite_from_params to work
with also which were already there just
Sorry this was meant for the Asterisk list. I wish FreeSWITCH had QSIG
support so I could go that route.
Ryan
On Mon, Sep 14, 2009 at 3:46 PM, Ryan Wagoner rswago...@gmail.com wrote:
I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make
calls from the Toshiba to Asterisk and
I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make
calls from the Toshiba to Asterisk and internal calls from Asterisk to
the Toshiba. What I can't do is make an call with an outside
destination from Asterisk to the Toshiba. The Toshiba is looking for 9
to grab an outside line
I seem to be missing something in implementing the ERLang callbacks
for Freeswitch. Our Freeswitch server is starting and getting
registered with ERLang, we're invoking the bind for configuration, but
I'm not seeing any of my callbacks fire. What am I missing?
Sample code follows:
Hi Michael,
You can count with me for anything else, like documentation,
coding/scripting, or any other FreeSWITCH related stuff.
Regards,
Diego
2009/9/14 João Mesquita jmesqu...@freeswitch.org
You can assign two things to me.
1. libesl code documentation (partially done and Doxygened -
I'm good for coming up with some documentation (which I'm doing anyway for
my guys at the office). Not that whats on the wiki isn't good and I'll
likely steal; its all there if you read it.
I'll submit this when I reach some measure of completeness. If its deemed
good, great. If not, well, I
Thanks Brian.
On Mon, Sep 14, 2009 at 8:55 PM, Brian West br...@freeswitch.org wrote:
You have a merge conflict please svn revert sofia.c
/b
On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote:
Hi Folks,
I've got a compilation error with the latest codes (r14842)
Making all in
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