Hi
Look at Contrib of source
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/
some pre-paid examples
Ram
On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi
senaka...@gmail.comwrote:
Dear Sir,
I have successfully installed freeSWITCH and it works fine in passthrough
Hallo All!
I need information about setup BLF on GXP2010/2020 phones with Freeswitch.
I search in Freeswitch Wiki and maillist archives but find no usable
information.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
I just found that this is related to the username of the profile. It needs to
be set as parameter.
Oscav wrote:
Hi,
Is it possible to set (rewrite) the Session Name in the SDP of a 183
progress sent to inbound ?
Many thanks
--
View this message in context:
Okay, I added: param name=apply-proxy-acl value=true/ to my sofia
profile and restarted sofia, and still no joy.
I'm on FreeSWITCH Version 1.0.trunk (15764)
I've got param name=auth-acl value=190.218.103.12/32/param in
the directory, but I'm still being rejected by the acl:
2009-12-17
Dear Ram,
Thank you for the reply. To work with your code I hope that Mod cdr
should be there. But wiki says that its not functional. What should I
do.
Thanks
Senaka
On Thu, Dec 17, 2009 at 7:29 PM, ram talk2...@gmail.com wrote:
Hi
Look at Contrib of source
with the scenario below can we get the better performance:
We create one profile for incoming call listening on 5060 as profile1
we create two profile for outgoing calls as profile2 on 5050 and profile3 on
5051
now we are receiving all calls on profile1:5060, but while bridging them to
vendors
I don't have access to a remote computer from which I could log on and run
nmap.
I'll see if I can get a shell access somewhere. Thank you.
--
View this message in context:
http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.html
Sent from the Freeswitch-users
Just for your information there is a version of nmap for windows. So you can do
the test from your desktop.
Оригинално писмо
От: Fred-145
Относно: Re: [Freeswitch-users] Scanning my firewall for open UDP ports?
До: freeswitch-users@lists.freeswitch.org
Изпратено на:
Hi Mike,
This has shown up on my laptop running ubuntu karmic. If we plan ahead, I
can setup ssh access for you to check things out.
In case this wasn't apparent I am trying to install FS from trunk.
Thanks,
Neil
On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris m...@jerris.com wrote:
strange,
it needs to be an ACL from acl.conf or a ip/cidr
/b
On Dec 17, 2009, at 5:41 AM, Bill W wrote:
Okay, I added: param name=apply-proxy-acl value=true/ to my sofia
profile and restarted sofia, and still no joy.
I'm on FreeSWITCH Version 1.0.trunk (15764)
I've got param name=auth-acl
Works on my CentOS 5.4 box just fine...
/b
On Dec 17, 2009, at 7:34 AM, Neil Patel wrote:
Hi Mike,
This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can
setup ssh access for you to check things out.
In case this wasn't apparent I am trying to install FS from
We need more info... svn rev, gcore, back trace and what not... please see the
reporting bugs link on the wiki.
http://wiki.freeswitch.org/wiki/Reporting_Bugs
/b
On Dec 16, 2009, at 11:53 PM, Juan Backson wrote:
Hi
I have rtp-timeout-sec set to 300 s but I am still getting calls with
Why are you needing to change it?
/b
On Dec 17, 2009, at 5:21 AM, Oscav wrote:
I just found that this is related to the username of the profile. It needs to
be set as parameter.
___
FreeSWITCH-users mailing list
I would be curious what the same tests produce with svn trunk of FreeSWITCH.
Mike
On Dec 16, 2009, at 4:49 PM, Brian wrote:
Hi,
I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to
see if it will scale better that other solutions. My scenario is to have one
Its software, anything is possible with enough time and effort.
Mike
On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote:
After some discussions with Polycom support it seems that their conferencing
support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the
latest and is
if you don't see it in sofia siptrace but do see it in tcpdump capture then
something very ugly is going on. Either sofia has hung up completely and is
not listening on that port anymore (can other calls go through?) or the packet
you see in tcpdump is not really going to the right port. Can
Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536
Please advise if you need further info.
Thank you.
From: Anthony Minessale anthony.miness...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, December 16,
It seems to me, in previous revisions of FS, we could successfully call a
registered user as soon as his terminal gets 200 OK for REGISTER.
But after testing recent revisions, it seems we must wait a little (I wait 1
second) otherwise a call to bridge would end with this:
2009-12-17
Is the packet capture running on the FS box itself?
On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris m...@jerris.com wrote:
if you don't see it in sofia siptrace but do see it in tcpdump capture then
something very ugly is going on. Either sofia has hung up completely and is
not listening on
if you contact me offlist, or better, join #freeswitch on irc.freenode.net and
ping me (MikeJ)
Mike
On Dec 17, 2009, at 8:34 AM, Neil Patel wrote:
Hi Mike,
This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can
setup ssh access for you to check things out.
In
The question was:
Are you doing the packet capture on the actual FS box using tshark or
tcpdump?
On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote:
Anthony,
I have pasted the invite sip trace here:
http://pastebin.freeswitch.org/11536
Please advise if you need further info.
Are you doing proxy or bypass meda?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 17-Dec-09, at 12:53 AM, Juan Backson wrote:
Hi
I have rtp-timeout-sec set to 300 s but I am still getting calls
with duration of 1 day
are you doing this trace from the freeswitch box itself?
Mike
On Dec 17, 2009, at 10:48 AM, DJB wrote:
Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536
Please advise if you need further info.
Thank you.
You could try ssldump:
http://www.rtfm.com/ssldump/
On Thu, Dec 17, 2009 at 12:16 AM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
Hello,
I am trying to debug a TLS handshake error between FreeSwitch and some
ATA. When setting the loglevel to 9 I get only a message that TLS handshake
I'll rephrase my question: Has anyone done that, or should I dig into it?
After all, Polycom is quite common...
Thanks, __Yehavi:
2009/12/17 Michael Jerris m...@jerris.com
Its software, anything is possible with enough time and effort.
Mike
On Dec 17, 2009, at 2:29 AM,
sip session timers is the standardized way to handle this.
On Thu, Dec 17, 2009 at 10:00 AM, Mathieu Rene mrene_li...@avgs.ca wrote:
Are you doing proxy or bypass meda?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
Hello Micheal
On Dec 15, 2009, at 12:09 PM, Michael Collins wrote:
Hi all,
What is the difference between the mod_voicemail vm_message_ext parameter
and the file-extension parameter?
vm_message_ext is a channel variable:
http://wiki.freeswitch.org/wiki/Mod_voicemail#vm_message_ext
Hello,
we are running freeswitch 1.0.trunk and are currently trying to get the
mod_voicemail to send the received messages to the user by using exim4 on a
debian machine.
So far we followed the instructions in the wiki article (
http://wiki.freeswitch.org/wiki/Mod_voicemail ).
I added
The sql is sorted into transactions to boost performance so it waits for
either 500 statements to execute or 500ms to elapse to accumulate as many
sql stmts as possible into the transaction.
set sql-in-transactions to false in your profile or make a patch to make the
500ms configurable
On
I'd be suspicious of:
(a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3;
(b) the branch on the Via tag changing
(c) (not sure about this one) the SDP session ID and version changing
for what's the same session.
--Dave
Anthony,
I have pasted the invite sip trace here:
What SVN rev. exactly?
/b
On Dec 17, 2009, at 10:13 AM, Oliver Schönbeck wrote:
Hello,
we are running freeswitch 1.0.trunk and are currently trying to get the
mod_voicemail to send the received messages to the user by using exim4 on a
debian machine.
So far we followed the
The trace that I pasted on the pastebin was from our analyzer,Tektronix
spectra2 that was sitting between FS and customer. I also had the FS sip trace
on and compare with the trace from Spectra when I found out about the 3rd
re-invite was missing from FS.
Thank you.
It only happened to the calls from this customer that keeps sending re-invite
every 30 minutes, since their switch is expecting a reply back from those
re-invite and FS did not respond back to those re-invite.
Thank you.
From: Michael Jerris m...@jerris.com
Currently it is Version 1.0.trunk (15982)
Von: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Im Auftrag von Brian
West
Gesendet: Donnerstag, 17. Dezember 2009 17:17
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users]
On Fri, Dec 18, 2009 at 1:12 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
The sql is sorted into transactions to boost performance so it waits for
either 500 statements to execute or 500ms to elapse to accumulate as many
sql stmts as possible into the transaction.
set
Hello
I'm no wget expert, and figured I should ask here first: I'd like to
download the whole wiki using wget for off-line reading.
Using the following didn't work:
wget -m -np http://wiki.freeswitch.org/wiki/Main_Page
If I move the wiki/ directory to the root directory of my web server, and
I would rather you not do that with wget you beat the hell out of the wiki
resources... how often do you do this? I would try doing a printable version.
/b
On Dec 17, 2009, at 10:56 AM, Fred-145 wrote:
Hello
I'm no wget expert, and figured I should ask here first: I'd like to
download
I only tried once. Maybe someone used to wget could generate a PDF in case
people need an offline copy?
--
View this message in context:
http://old.nabble.com/Mirroring-wiki-with-wget-for-offline-browsing--tp26831043p26831566.html
Sent from the Freeswitch-users mailing list archive at
On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote:
Hi,
I'm probably going to regret this - I'm not sure that I'll be able to do this
without a lot of pain (nothing to do with FS - more my lack of ability with
Visual Studio), but.., I want to try building FreeSwitch from
Anthony/Michael,
I finally have a complete traces of a call at
http://pastebin.freeswitch.org/11539
There are 2 traces in there from within the same box. One is from
freeswitch/sofia siptrace debug and the other one from ngrep for your
comparison.
The missing re-invite in FS is at
Hi,I am trying to define Gateways (for inbound and outbound calls via SIP
provider) within Directory (under internal sample profile) using XML CURLBut
I am getting this warning:2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115
Gateway 'MyGW' not found.
And
sofia status gateway MyGWAPI CALL
I have not seen anyone mention it.
Mike
On Dec 17, 2009, at 11:07 AM, Yehavi Bourvine wrote:
I'll rephrase my question: Has anyone done that, or should I dig into it?
After all, Polycom is quite common...
Thanks, __Yehavi:
2009/12/17 Michael Jerris m...@jerris.com
I'm going to guess you removed these lines from your profile:
domains
domain name=all alias=false parse=true/
I am sorry; here is the complete one: http://pastebin.freeswitch.org/11540
Thank you.
From: DJB djbin...@yahoo.com
To: freeswitch-users@lists.freeswitch.org
Sent: Thu, December 17, 2009 9:35:27 AM
Subject: Re: [Freeswitch-users] SIP Re-invite
Hi,
If I configure data as following, why FS A 1001 call FS B 1003 failed ?
Thank you!
FS A: 192.168.129.168, DN=1001
FS B: 192.168.129.194, DN=1003
In FS A add /conf/sip_proifles/external/gwfsa.xml
include
gateway name=gwfsa
/gateway
/include
1101 is
Can you post the full packets with Ethernet, IP, UDP headers as well, or
upload a pcap file?
I'll add the change in 'Max-Forwards' from 70 to 69 between the two
packets to my things to be suspicious of list.
--Dave
The trace that I pasted on the pastebin was from our
analyzer,Tektronix
I found the issue with this. I did an svn checkout from the trunk, and then
I did a local svn export to another local folder. For some reason, the svn
export did not include the libs/openzap folder (which was not the case when
I got 1.0.5pre8). Must I do a separate svn export from the
On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote:
On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote:
Hi,
I'm probably going to regret this - I'm not sure that I'll be able to do
this without a lot of pain (nothing to do with FS - more my lack of ability
with Visual
This would have nothing to do with receiving a 502 on sip.
/b
On Dec 17, 2009, at 12:08 PM, Jerry Richards wrote:
I found the issue with this. I did an svn checkout from the trunk, and then
I did a local svn export to another local folder. For some reason, the svn
export did not include
Hello everyone,
I've got two profiles running: s2s and trunk. The context for s2s is
defined as s2s-in. The context for trunk is defined as trunk-in.
trunk is bound to 192.168.168.3.
recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706:
Please try on SVN trunk. I might toss a PRE10 sooner.
/b
On Dec 17, 2009, at 1:05 PM, Juan Backson wrote:
Hi,
I am using 1.0.4 (exported) with proxy media, and I have enable-timer = true
and minimum-session-expires=120.
Is this the correct way of setting the sip session timers?
Hi Mike,
I didn't get around to testing on the FreeSWITCH trunk yet. Are there
substantial fixes to mod_conference in the FreeSWITCH trunk that might
increase capacity for my scenario of one speaker and many listeners? If I
want to put this into a production environment, I would need a stable
If you're going to have that many listeners then it would be best to use
something like shoutcast to broadcast the stream out to a local stream on
various different boxes... then tie the callers into a stream... when they have
questions uuid_transfer them into the conf.. then back to the stream
I am trying Audiocodes and Vegastream ATAs, and work with either the
manufacturer or the local representative here.
On SNOM I managed to make it work, and will try Polycom soon (once I manage
to grab one unit from our users...).
Thanks, __yehavi:
2009/12/17 Brian West
One man's stable release is another man's 6 month old release with hundreds
of known fixed bugs.
If one of the core developers tells you to try it, you may as well take the
time to try it now that you have opened a forum questioning the scalability.
When you tested asterisk did you actually use
We are always doing enhancements and yes there are some real scalability
enhancements in trunk compared to 1.0.4, I am just not sure if they effect
conference significantly or not. I would guess that trunk is actually more
stable than 1.0.4 at the moment. Give it a try and find out.
Mike
On
The sounds projects (which do the downloads and extraction) are not present
for 2005. Also alot of the newer modules dont have build support either.
I would suggest you use VS2008 Express
Michael Jerris wrote:
On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote:
On Thu, Dec 17, 2009 at
I run FreeSWITCH on a Windows Server 2008 R2 (x64) box with several analog
lines and it works very well.
mercutioviz wrote:
And we shouldn't be using 1.0.4 anyway, should we? ;)
-MC
On Wed, Dec 16, 2009 at 3:26 PM, Moises Silva
moises.si...@gmail.comwrote:
I've been using
Yes, while it is true that does make a profound difference but if he has many
listeners and not very many talkers... just tapping into the conference and
streaming that audio out would scale well.
/b
On Dec 17, 2009, at 1:50 PM, Steve Underwood wrote:
I don't think you have mentioned which
I didn't realize there was a policy about load testing questions. What forum
should I have used for this?
I didn't get the chance to test on FS trunk yet, but when I do I will
provide you with the feedback when I do. Just let me know what forum to use
for this topic from now on.
Thanks,
Hi,FS was sending (while loading modules) such request: [purpose] = gateways
But I was not aware of that...so that I am replying FS with my Gateways now...
But now I am wondering...suppose I have 1000 domains and two different gateways
per domain (2K Gateways) Should I reply FS request with
We didn't post it anywhere but we just get overwhelmed with them and many of
them are unfounded and take up a lot of time to track down. That does not
mean you have not found a real problem but the first step is trying trunk.
On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com
Hi Brian,
I imagine that one of the issues is that you're using a complex
sledgehammer (mod_conference) to crack a simple nut - that of having
multiple listeners listening to a single speaker.
As far as I am aware, FreeSWITCH doesn't have anything built in which
will allow this kind of simple
In your case don't store them in the domain put them in the gateways tags on
the profile directly.
/b
On Dec 17, 2009, at 2:46 PM, Paulo Vicentini wrote:
Hi,
FS was sending (while loading modules) such request: [purpose] = gateways
But I was not aware of that...so that I am replying FS
I try to attach Bravis video conference clients to Freeswitch. Those
video conference clients are really working good (Multilingual clients
for testing ca be downloaded here: http://www.bravis.eu/). Some big
companies here in Germany use them in large installations. They are
based on SIP, but do
Looking at Performance Tune my Freeswitch
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
Is refers to the following:
Turn off every module you don't need
Turn presence off in the profiles
libsofia only handles 1 thread per profile, so if that is your bottle neck use
1. http://wiki.freeswitch.org/wiki/Modules.conf.xml
http://wiki.freeswitch.org/wiki/Modules.conf.xml2.
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence3.
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#SIP_Profiles
libsofia only handles 1 thread per profile, so if that is your bottle neck
use more profiles
If you only have one provider for your trunk is it possible to set up
multiple profiles for enhanced performance?
For example if I have multiple DDIs from the provider can I set up a
different profile
Hello Oliver,
I have the same on Ubuntu wth newest trunk.
Best regards
Peter
Oliver Schönbeck schrieb:
Hello,
we are running freeswitch 1.0.trunk and are currently trying to get
the mod_voicemail to send the received messages to the user by using
exim4 on a debian machine.
So
What exactly are you doing I know it goes better than that.. are you using
64bit?
/ b
On Dec 17, 2009, at 3:41 PM, Brian wrote:
I did a test with the trunk version for the one conference case, and it is
the same results as for 1.0.4. The audio failed at around 300 listeners.
Oddly though,
yah it's exim segfaulting because you have to configure it to emulate
sendmail per the wiki page.
On Thu, Dec 17, 2009 at 4:17 PM, Peter P GMX prometheus...@gmx.net wrote:
Hello Oliver,
I have the same on Ubuntu wth newest trunk.
Best regards
Peter
Oliver Schönbeck schrieb:
Hello,
What exactly is your test process?
you should try increasing the interval in the conference profile to a bigger
time slice maybe 30 40 or 60ms
you could also increase the ptime to match as well.
like brian said you could use mod_shout to broadcast the single speaker to
icecast and let people
I bit off topic but.
Using FS to send calls sip to the LD carrier.
Some calls have problems where they drop the call or audio drops or
whatever.
The carrier's first response is that we dropped the call. But this is
a day later after the trouble has been reported.
I am looking for guidance
Hey Brian,
I've been doing some testing and I am unable to get auth-calls to work
through a proxy the way I want them to, even with setting
apply-proxy-acl to either the endpoint IP or the proxy IP.
I have a multi-tenant system with multiple domains with multiple users
in each domain. And I
Thanks for the hint!
force_transfer_context and force_transfer_dialplan.
I've updated the wiki (I'll add an example once I test it).
On Thu, Dec 17, 2009 at 5:06 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
The calls inherit the context from the parent, I think there is a var you
Frank,
Probably the cleanest (albeit non-FreeSWITCH) way to implement this
would be to use OpenSIPS/SER/etc between you and the carrier with the
siptrace module.
But that's probably more work than you want. There's always tcpdump
with a decent filter (udp port 5060 and host x.x.x.x) and
On Thu, Dec 17, 2009 at 3:59 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Thanks for the hint!
force_transfer_context and force_transfer_dialplan.
I've updated the wiki (I'll add an example once I test it).
I love it when users go all Chuck Norris and Rambo in answering
On Thu, Dec 17, 2009 at 4:01 PM, Frank @ Impact fr...@impactfax.com wrote:
I bit off topic but…
Using FS to send calls sip to the LD carrier.
Some calls have problems where they drop the call or audio drops or
whatever.
The carrier’s first response is that we dropped the call. But
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
Also when can we expect little KK's running around? :P Congrats on the
marriage
/b
On Dec 17, 2009, at 6:27 PM, Michael Collins wrote:
I love it when users go all Chuck Norris and Rambo in answering their
questions AND documenting the info! Thanks KK.
-MC
So is wireshark UI and its free! :P
/b
On Dec 17, 2009, at 6:33 PM, Chris Fowler wrote:
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…
___
FreeSWITCH-users
i agree with christian, though i would use tshark. you can actually
get the fields you want (method and callid) and store them in a dB.
then you need to match them with a query. it is simple but Lots of work.
look into -e and -E of tshark separate the fields by ,
have fun!
David
El
I'm using contrib/seven/sip/sip2db.rb
2009/12/18 David Villasmil david.villasmil.w...@gmail.com:
i agree with christian, though i would use tshark. you can actually
get the fields you want (method and callid) and store them in a dB.
then you need to match them with a query. it is simple but
I couldn't guess what you want, pastbin your full config and logs and
give more detail of your story perhaps someone can help you.
2009/12/18 yvonne ding yhding2...@yahoo.ca:
param name=username value=1101
param name=password value=1234
param name=proxy value=192.168.129.194:5060
param
Please check your dialplan to match the new extension.
You are looking for dialplan/default.xml extension Local_Extension. Check
the cond destination_number, it should give you a good hint.
Regards,
JM
On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz darklio...@yahoo.com wrote:
Hi Sir,
I
Some providers do retain call data for diagnostic purposes and to to aid
in troubleshooting. Why not politely ask them if they could provide you
with a sip trace themselves or forward along the evidence that supported
their conclusion. They should be willing to help you solve a problem
that
Hi Sir,
Not working condition field=destination_number
expression=^(10[01][0-9])$ i set this to
condition field=destination_number expression=^(80[1][0-9])$ to
call 801.xml up to 809.xml on the dialplan/default.xml same thing...
Thanks,
Edmar
João Mesquita-4 wrote:
Please
Hey Metik,
Thanks for the reply, and the pointers for doing it with xml_curl.
I'll guess have to do that in the short term, but in my opinion, having
auth-acl be able to work through a proxy is very important as it is a
vital part of a comprehensive security feature set. And it would be
much
Why not simply implement this feature in the PROXY itself?
FS has a pretty comprehensive security feature set for endpoints that
directly register with it.
Don't get me wrong, I do agree this is useful especially if you are
going to be using your proxies to load balance across multiple FS
From looking at sofia.c, if the ip address of the caller is in apply-
proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet,
and use that one for authentication.
Is that what you did in your previous tests?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Hey Metik,
That's exactly what I'm trying to do... load balance across multiple FS
boxes, and have any machine in the cluster be able to reach a device
behind a NAT firewall. Hence the need for the proxy. Also, I'm trying
to keep the proxy relatively dumb and put all the logic in the FS
Hi Everyone,
Is there a link or tutorial for the expressions format.
Example:
condition field=destination_number expression=^(10[01][0-9])$
10 - default number
[01[- second number that start only on 0 or 1;
[0-9] - 0 to 9 can be use
Is there
Hi All
I have a FS cluster behind two OpenSIPS proxy, the incoming calls is load
balance and failover to FS cluster by OpenSips, It works well.
The problem is, the outbound calls from FS must also route throw then
OpenSIPS servers. So, does FS servers can loadbalance the outbound calls
between
On Dec 17, 2009, at 11:34 PM, Jason White ja...@jasonjgw.net wrote:
Edmar Cruz darklio...@yahoo.com wrote:
Is there a link or tutorial for the expressions format.
Anything that describes Perl regular expressions should help, and for
reference, see the pcre(3) manual page, and use the
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