Dear All,
What is the value of created_time channel variable? Is this epoch
seconds?
Thanks Regards,
Velusamy.
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Yes, I can confirm - this exact error occurs each time when I start recording
before the call is answered (just after sending ORIGINATE command) - but I
think that's completely understandable that media is not ready for an
unanswered call.
But... is there any other event that guarantees media to
P.S. people from russian community report what current version of module
work fine on fs
trunk version.
that's strange that they report it working as m_txAudioOpened is never
gonna be ready in pre_answer :P... i had to comment it to make it working.
anyhow, i moved everything to trunk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
hm, no not really. There is no change in this behaviour in FreeSWITCH
Version 1.0.trunk (15225M)
There is still a caller name change in callee's display.
I'm not sure who is wrong here. Either FS or Snom ...
regards
Helmut
On
Dear All,
I'm running mod_nibblebill for my prepaid solution. I
still have problem with memory. I have 4 GB RAM and runing debian
squeeze 64 bit and 200 calls concurrent
Last time nibblebill running with 1 min heartbeat. when i
check memory by htop FS user memory 2% anf
Thank you for the get idea, It works perfectly.
On Fri, Oct 23, 2009 at 4:37 PM, Metik freeswitch-users-l...@metik.comwrote:
Why not simply overwrite the value of the variable used throughout the
script...
-- xml_curl.conf --
...
param name=gateway-url value=
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
thx, it works now again. But the only way get it to work is using
send_display after pre_answer. (sip|originate)_callee_id_name doesn't
work with playback.
regards
Helmut
On 23.10.2009 16:33, Anthony Minessale wrote:
There was a
No you can't remove them... And they are 100% valid so your SBC is in
the wrong.
/b
On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:
Hi,
I used the downlaoded TAR ball and my calls worked, however, when
upgrading to the SVN release...my SBC is rejecting the 200 OK (when
the FS
Hi All:
I'm a newbie to FS. I'm using FS as a sbc and have about 2 user
account . Does somebody can tell me how to make FS load use account
information from a database such as mssql or mysql? Could you give me a
sample configuration file?
Thanks a lots.
Could you maybe consolidate all of your problems into 1 thread. I am
getting dizzy. You have 2 on the same subject and you say it works on one
and does not on the other.
Last week we tested all of this with latest trunk and there is no longer any
problems of any sort with the display related
I originally sent this last Friday but I've been unable to confirm it
ever made it to the list.
Hello everyone,
I'm having some issues with SIP and TCP. I've used it before with
success but I'm seeing some strange behavior...
Level 7 debugs with siptrace on both profiles. UDP invite from
Lei,
I am still learning myself, but I think I can help enough and others can
chime in where I am wrong.
If you have FS up and running you will need to install xml_curl (
http://wiki.freeswitch.org/wiki/Mod_xml_curl).
Then you will need to edit the config for it in your
It was never there before and it caused extreme havoc once we added it so we
took it away again.
On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote:
I am wondering why after I update to trunk-15225, the Allow: UPDATE method
is no longer there.
User-Agent:
Brian,
This bothers me a bit.
Of course they are valid. Anything prefixed with X- should be
ignored by the remote end unless they are specifically looking for it.
However:
1) We all know that just because the spec says they are valid doesn't
make it so with every vendor.
2) As long as
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1...@uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the
i cant seem to reproduce it.
originate sofia/internal/1...@conference.freeswitch.org;transport=tcp 9998
I get a working call and trace.
Could you possibly have a dns error? I know it's an ip but it may still
fail if it has no dns.
try
sofia loglevel all 9
and look for other errors.
On
Hello VoIP Geeks!
I am looking for a way to reroute calls on specific disconnect reasons. My
application is in Python. I will provide my solves so far, which for some
reason I cannot use. Why I need rerouting? Well, I am using the LCR module
which provides multiple routes per destination, like
On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov shiya...@gmail.com wrote:
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
With the following spec for CPU and Memory can someone help me guesstimating
how many simultaneous calls and Calls/sec a FS server can handle - Used as a
Conferencing Server.
cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 4
model name
The headers are used to pass the callee-id info back to the other side so
you have the id of who you called.
The standards have failed us in this case as everything does it differently
to the point that there is no standard thus we have invented our own way to
carry this across from one FreeSWITCH
Tony,
It seemed strange to me too (I'm using TCP in other places).
I'll take another look at this with your suggestions for debugging.
Thanks!
On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
i cant seem to reproduce it.
originate
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
sorry for making you dizzy ... bun in fact in my point of view I have
two different problems.
1.
One concerns the way using send_display in pre_answer mode. Simply to
send error texts to caller's display. This works again with latest
Tested- it works!
Thanks a lot!!
On Mon, Oct 26, 2009 at 6:32 PM, mayamatakeshi mayamatake...@gmail.comwrote:
On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov shiya...@gmail.comwrote:
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
Anthony,
So if I'm understanding you correctly, if you are always using
FreeSWITCH as an edge to other systems you should be able to safely
disable these headers?
On Mon, Oct 26, 2009 at 11:41 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
The headers are used to pass the callee-id
depending on your dialplan every time you bridge to a channel it changes the
display to match who you are talking to. if you tried to set it with the
variable then you call someone that is going to cause this. Take away the
display app and/or any special variables and let it naturally work.
On
On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
The headers are used to pass the callee-id info back to the other side so
you have the id of who you called.
The standards have failed us in this case as everything does it differently
to the point that
Thus perpetuating the wild-west of sip where you can't do anything according
to spec because you have to worry about stupid things not keeping up.
Sounds like the education system where I live too.
I'll see what I can do. It's always the other end that ppl pay for that
drive the free stuff
I'm also having problems with this. When running FS compiled about 10 days ago
it works fine (don't remember exact revision), but when using latest SVN it
doesn't work anymore. I seems like it's trying to use UDP when it should use
TCP.
My setup is this: Avaya Communication Manager PBX - Talks
Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to
the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad!
However, something that has changed the last 10 days seems to affect my setup
so it doesn't work anymore. I'll do some more SIP tracing, and get back
You'll have to do your own load testing. Nobody can really tell you
exactly how many you'll get.
/b
On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote:
With the following spec for CPU and Memory can someone help me
guesstimating how many simultaneous calls and Calls/sec a FS server
Finding the exact rev that broke it would be helpful.
/b
On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:
Hmm... I remembered incorrectly about my setup :) The Avaya PBX
talks TLS to the Avaya SES Server, and then UDP to FS, not TCP -
sorry, my bad!
However, something that has changed
Yes, I know :) However, now I think this is related to the new headers
introduced, it's probably not a TCP issue.
Everything seems to work just fine until the 200 OK is sent, the Avaya PBX
doesn't seem to accept that reply anymore.
The only differences I've found between a working revision,
Bet your hardware just barfs on those like others have... I mean
really I HATE SIP. This is stupid.
/b
On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:
In the non-working one I don't have these, and instead I have these
headers;
X-FS-Display-Name: 9099
X-FS-Display-Number: 9099
Get a dedicated DSL line. They aren't that expensive... I have four
of them at my house!
/b
On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:
Is there a write-up anywhere that might help me with this problem,
or lacking that, can anyone offer advice?
]. failure #1
Thanks Lars
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signature database 4545 (20091026) __
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http://www.eset.com
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I agree, it might be a dirty solution, but its so much easier than
trying to get QoS running on a DSL line, or learning how to run a
traffic classifier...
On Mon, Oct 26, 2009 at 5:46 PM, Brian West br...@freeswitch.org wrote:
Get a dedicated DSL line. They aren't that expensive... I have four
try r15230
add the profile param
param name=pass-callee-id value=false/
On Mon, Oct 26, 2009 at 12:46 PM, Brian West br...@freeswitch.org wrote:
Bet your hardware just barfs on those like others have... I mean
really I HATE SIP. This is stupid.
/b
On Oct 26, 2009, at 12:39 PM, Peter
This is ridiculous but here it is
try r15230
add the profile param
param name=pass-callee-id value=false/
On Mon, Oct 26, 2009 at 11:16 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Thus perpetuating the wild-west of sip where you can't do anything
according to spec because
I understand your frustration :) We deal with SIP integration with about 10
different PBX vendors today, And it's always something that doesn't work as it
should. Right now I don't have anything more connected to FS though.
/Peter
-Ursprungligt meddelande-
Från:
New sofia profile param as follows:
!-- set this param to false if your gateway for some reason
hates X- headers that is is supposed to ignore--
!--param name=pass-callee-id value=false/--
On Oct 26, 2009, at 12:16 PM, Anthony Minessale wrote:
Thus perpetuating the wild-west of sip
At some point we'll have to NO NO NO fix your broken crap. :P The
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with
me... NO!
/b
On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:
I understand your frustration :) We deal with SIP integration with
about 10 different
On 26/10/09 10:37 -0700, Lars Zeb wrote:
If one of the computers does a big download, it messes with FS in two ways.
If a connection is made, the voices are broken up, intermittent and
difficult to understand. If the download is long enough, the connection to
Flowroute is no longer usable due to
what exactly are you expecting to use it for?
We never really supported it anyway.
On Mon, Oct 26, 2009 at 12:19 PM, DJB djbin...@yahoo.com wrote:
I wonder whether you will consider to put it back on the next version 1.0.5
since 1.0.4 has it?
Regards,
Dorn B.
We were trying to see whether we can adjust call duration on Session timers.
It was a question from the application developers. I am not sure what they are
trying to do exactly.
Thank you.
From: Anthony Minessale anthony.miness...@gmail.com
To:
2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute Registration
Failed with status Operation has no matching challenge [904]. failure #1
Thanks Lars
__ Information from ESET NOD32 Antivirus, version of virus signature
database 4545 (20091026
I think session timers will use invite if there is no update.
the session-timeout profile param should control that but you have to double
the number you actually want because it sends the new invite at the halfway
point.
On Mon, Oct 26, 2009 at 2:29 PM, DJB djbin...@yahoo.com wrote:
We were
but i think the minimum value you can set is 120
On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I think session timers will use invite if there is no update.
the session-timeout profile param should control that but you have to
double the number you
Are there any benchmarking test results available publicly?
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
[br...@freeswitch.org]
Sent: Monday, October 26, 2009 11:18 AM
To:
in r15233 i put it back to the way it originally was but I may have to
remove that if it causes more problems.
We tried to handle update for display updating which was not working but the
handler for it was still in place which may have broken some automatic
behavior regarding update so I added it
On Mon, Oct 26, 2009 at 11:40 AM, Brian West br...@freeswitch.org wrote:
At some point we'll have to NO NO NO fix your broken crap. :P The
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with
me... NO!
/b
I was wondering... does anyone make a SIP certification program
I highly doubt it... You can wait for someone to post their results
but in the end you'll have to do your own load testing because not
everyone's numbers will jive with your use case. Which is the reason
the project never posts or endorses a set call count.
/b
On Oct 26, 2009, at 2:50 PM,
You have SIPit, which was the SIP Backoff till Pillsbury got their
panties in a wad.
/b
On Oct 26, 2009, at 3:03 PM, Michael Collins wrote:
I was wondering... does anyone make a SIP certification program
kinda like a pen-tester except to find all the ways your SIP setup
is broken? Just
On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi All:
I'm a newbie to FS. I'm using FS as a sbc and have about 2 user
account . Does somebody can tell me how to make FS load use account
information from a database such as mssql or mysql? Could you give me a
Here are a few benchmarks that I had stumbled upon.
http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
Thanks,
Vinuth.
On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org
It is ridiculous but thank you very much!
On Mon, Oct 26, 2009 at 2:26 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
This is ridiculous but here it is
try r15230
add the profile param
param name=pass-callee-id value=false/
--
Kristian Kielhofner
http://www.astlinux.org
Woof!
On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins m...@freeswitch.org
wrote:
I was wondering... does anyone make a SIP certification program kinda
like a pen-tester except to find all the ways your SIP setup is broken?
Just curious.
Here is a start:
http://interop.sipxecs.org/
On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
vinuth.madi...@gmail.com wrote:
Here are a few benchmarks that I had stumbled upon.
http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
Please remember NO benchmarks are endorsed by the FS community or
developers,
to register which looks worrying:
http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png
I'm pretty sure that I'm not filtering out the emails but can't be sure.
Could whoever maintains the web page contact me off list to help me
determine where the problem is?
Thanks,
Simon Mudd
On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz
maciej.aniserow...@gmail.com wrote:
Yes, I can confirm - this exact error occurs each time when I start
recording
before the call is answered (just after sending ORIGINATE command) - but I
think that's completely understandable that media
On Fri, Oct 23, 2009 at 10:31 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
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
with the
same results, no mail received.
I also see the following when trying to register which looks worrying:
http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png
I'm pretty sure that I'm not filtering out the emails but can't be sure.
Could whoever maintains the web page
On Mon, Oct 26, 2009 at 7:26 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
This is ridiculous but here it is
try r15230
add the profile param
param name=pass-callee-id value=false/
sorry for that but, this will save you a lot of e-mail explaining why calls
are not going
If one of the computers does a big download, it messes with FS in two ways.
If a connection is made, the voices are broken up, intermittent and
difficult to understand. If the download is long enough, the connection to
Flowroute is no longer usable due to registration failure.
In any
Thanks for your response Michael,
Both the resources you've referred to don't explicitly say much about
databases. Could you elaborate just a bit on how on information would
retrieved from a database (MySQL) and presented to mod_xml_curl.
Thanks again.
2009/10/26 Michael Collins
Unfortunately, most North American providers ignore (and in most cases
reclassify) it before it reaches their border routers and it will be treated as
best effort. Typically, the problems are introduced within the first mile and
its simply a matter of getting the packet safely pass the edge
Although, FYI, I just benchmarked mod_xml_curl on a separate web app
server from FS with FS on a Dell R710 with their current best
processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
GB memory. The web app server is less than half the power of the R710.
I maxed the web app
i wonder if we can at least get a taco-bell steak burrito for that if we
can't win the s-prize
On Mon, Oct 26, 2009 at 5:01 PM, Eliot Gable
egable+freeswi...@gmail.comegable%2bfreeswi...@gmail.com
wrote:
Although, FYI, I just benchmarked mod_xml_curl on a separate web app
server from FS with
On Mon, Oct 26, 2009 at 4:03 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
i wonder if we can at least get a taco-bell steak burrito for that if we
can't win the s-prize
Or at least a chalupa.
-MC
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Why don't you get us more information for debugging? We could use some vg
output, maybe?
JM
On Mon, Oct 26, 2009 at 9:24 AM, Dome Charoenyost d...@tel.co.th wrote:
Dear All,
I'm running mod_nibblebill for my prepaid solution. I
still have problem with memory. I have 4 GB RAM and
yo quiero taco bell
/b
On Oct 26, 2009, at 7:04 PM, Michael Collins wrote:
Or at least a chalupa.
-MC
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I have tried to update (make current) twice since 15183. All inbound calls
are picked up but the caller hears nothing but a couple of clicks. The most
recent version I've tried is 15241.
Any ideas on what may be causing this?
http://pastebin.freeswitch.org/10843
Linux fs
you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip
correctly?
/b
On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote:
I have tried to update (make current) twice since 15183. All inbound
calls are picked up but the caller hears nothing but a couple of
clicks. The most recent
FYI, it generally makes debugging easier if you do this:
sofia profile external siptrace on
sofia profile internal siptrace on
That way you can see the actual signaling and it is usually more clear
what is going on. In most cases, you will probably be able to figure
it out yourself just looking
I haven't changed anything since v15183, where it worked OK.
In conf/sip_profiles/external:
param name=ext-rtp-ip value=auto-nat/
param name=ext-sip-ip value=auto-nat/
And
Nameexternal
Domain Name N/A
DBName
Hi all,
I've done experimenting with the uuid_displace and mux. What mux does is
playing a file when the conversation is also happening. But I've a different
requirement.
I need to play a background music to a UUID, that will get played
continuously and also I need to play some other voice
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