Hold is working fine I just tested it... I would need to see the whole dialog
to see what is wrong... I tested with Polycom, Snom and Aastra.
Are you doing proxy media or anything like that?
/b
On Dec 29, 2009, at 1:14 AM, Lei Tang wrote:
Hi, I think hold function in trunk 16055 is broken,
Also can you join #freeswitch-dev, include full siptrace+debug log and put it
on pastebin.
What phone are you using?
/b
On Dec 29, 2009, at 1:14 AM, Lei Tang wrote:
Hi, I think hold function in trunk 16055 is broken, I have also tried some
old trunks, it's ok in freeswitch 1.0.4.
The
the 200ok is not from FS.. its from the end point... so its not us thats not
putting the SDP into the 200ok but the device you're talking to because in
proxy media they are passed as is.
/b
On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
Hi Brian, thanks for your help, I am using FS in proxy
Its null because the device on the other side didn't send one. We pass it as
is... fix the broken device or don't use proxy media.
/b
On Dec 29, 2009, at 9:37 AM, Lei Tang wrote:
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok
Ivan,
I have been trying to gather up everyone to start a FreeSWITCH based
softphone project for Mac, Linux and Windows... you think we could collaborate
with you to accomplish this? I think if we do this right we can have a really
nice phone with lots of options.
Thanks,
/b
On Dec
I would love to have a FreeSWITCH based softphone for all three platforms... I
just feel a project like that would be kick ass.
Must work on 32bit and 64bit of Windows, Mac and Linux ... and not suck like
most softphones do.
/b
On Dec 29, 2009, at 2:08 PM, EdPimentl wrote:
Add me to the
param name=media-option value=resume-media-on-hold/
But it doesn't go back to bypass after Maybe you can post a bounty for
that functionality.
/b
On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote:
When I uncomment the following tag, internally held calls no longer hear
MOH.
param
Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since
the libiax2 from Digium is no longer updated to keep pace with Asterisk and is
now incompatible. Which is the main reason we are thinking about dropping IAX
support unless someone writes a license compatible lib or
29. des. 2009 kl. 22.14 skrev Brian West:
Does it only do IAX? If so we'll need someone to re-write an IAX2 stack
since the libiax2 from Digium is no longer updated to keep pace with
Asterisk and is now incompatible. Which is the main reason we are thinking
about dropping IAX support
Anyone have access to these phones? Two of them if possible and provisioning
information?
Thanks,
Brian
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Sounds like a plan to me... who wants to take the lead on the project... we'll
host it.. setup SVN, provide jira access, fisheye and wiki space...
/b
On Dec 29, 2009, at 6:44 PM, João Mesquita wrote:
Why don't we evolve FSGui to be a softphone? I could use a couple of
experienced
Amplify Query... not enough data to make a logical compilation of requested
data.
/b
On Dec 30, 2009, at 12:17 AM, Sharad wrote:
Hi
I just want to know what should be the approx configuration of the server
for 50 concurrent call sessions having 3000-4000 users.
Regards
If you're using the 401 as an indication that it fails then you don't
understand how digest authentication works. I would have to see what happens
after the 401 to see if it really did fail.
/b
On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote:
This is all I see and then registration
Shared will require some testing with TLS. I need traces, console logs and you
to do some foot work to see if you can provide more details.
/b
On Dec 24, 2009, at 8:35 AM, Yehavi Bourvine wrote:
Hello,
Is there anyone who is using SNOM with TLS encryption and shared lines and
it
Dear FreeSWITCHers,
Someone has registered the freeswitch name and is squatting on twitter
with it. They haven't used it in over a year and I would like to have this for
our project as its clearly confusing.
If you own this account please contact me off list.
Thanks,
Brian
I'm still not done with this I think we found a bug in the lib... Viktor fixed
it today and I'm going to retry after I get done testing G729 more today! ;)
/b
On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote:
Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn trunk,
VMD will force a transcode anyway too.
/b
On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote:
My setup is as follows:
FreeSWITCH - SIP Trunk - PSTN.
From freeswitch, I'm making outbound calls using event socket via the
external profile. Except for the ext_rtp_ip and ext_sip_ip,
What does pretty much mean to you? Can you give me an exact rev?
/b
On Dec 23, 2009, at 8:26 AM, TTNC - Technical wrote:
Oh, I'm running pretty much the latest svn truck.
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Yes DNS is required for this to work properly.
/b
On Dec 23, 2009, at 9:43 AM, John wrote:
Still having this issue. Do separate domains need to be real fully
qualified domains, or can they just be added as in Company1, 2, 3, etc?
___
That usually means they are saying 30 but sending 10 which is broken.. you
can't say hey i'm sending 30 and then send 10... find a new provider or beat
them to death with a cluebat in hopes they fix their broken stuff.
/b
On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use the SIP
You might also have to set the codec negotiation to scrooge
/b
On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote:
You can disable auto-adjust in the sip profile., but that might just make it
worse, no warranty:
param name=rtp-autofix-timing value=false /
Mathieu Rene
Avant-Garde
2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by
acl 192.168.10.0/24[]. Access Granted.
Because the context is set on the profile as public... and you really didn't
auth the user so user_context was never set.
/b
On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote:
I am
The force-register-domain and force-register-db-domain are set in the defaults
so you can only do one domain. Remove those and you'll be able to do multiple
domains.
/b
On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote:
I have Freeswitch setup and working as a single tenant
system mostly
I'm not too sure you can put an a1-hash on outbound auth.
/b
On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote:
Hi,
Does any body know or has test the a1-hash parameter with gateway
setting? I am not sure if it is even allowed. I have the following
gateway setting but when the
Why? You don't have to avoid it... why bother?
/b
On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote:
My basic intent is to avoid on-the-fly transcoding, while having a high
quality audio playing on PSTN.
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If its degrading like that you have bigger issues... the sound files played
from wav files vs raw PCM files is NO different on a land line and I speak from
very many years of experience... your wav files are ulaw in wav containers thus
will never play native which might just be part of your
Can you get me siptraces please.
/b
On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote:
Thanks Brian and Gad,
I have stun set and if I do a 'sofia status profile internal', I see
the external IP address of the 3102 ATA, so I assume that stun is
working correctly on the SPA3102.
So says the man with his Skype username in his sig! :P
/b
On Dec 21, 2009, at 12:37 PM, Itamar Reis Peixoto wrote:
the best answer is don't use skype.
Itamar Reis Peixoto
e-mail/msn/google talk/sip: ita...@ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063
You'll need to fix your device to know its IP and it should stop doing that.
/b
On Dec 20, 2009, at 5:58 AM, Mark Campbell-Smith wrote:
Hi!
I'm sure this is a NAT issue, but I'm not sure what options to use.
I have a Linksys SPA3102, NAT'd on the internet (remotely) and
connected to my
On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote:
DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a
grain of salt.
Welcome to the community.
I have a similar setup (and problem) - the wiki documentation refers to it as
double nat. Like you, my FS and client are
You have to watch it with TLS. Make sure your distro didn't mess up your SSL
libs due to the recent vulnerability found. I havn't tested with my polycom in
a few weeks but it was working on my Polycom after I uploaded the ca cert and
marked it as trusted/used on the phone.
/b
On Dec 20,
The funny part is... it won't matter. Their are times when people post
questions or issues and its well into debugging the issue before we realize
oh, you're on windows?. For the most part the windows installer is one of
the most popular files on our website.
/b
On Dec 19, 2009, at 10:18
That depends if the call is answered and then you transfer it, you will HAVE to
set the transfer_ringback variable you can't send a 180 to the thing or a
progress and make it generate the ringback. You MUST do it yourself.
You also fail to mention if the progress is a 180 or a 183 with sdp and
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.
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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
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'm going to guess you removed these lines from your profile:
domains
domain name=all alias=false parse=true/
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
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?
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
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
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
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,
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…
___
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use apply-proxy-acl on the sofia profile.
/b
On Dec 15, 2009, at 10:58 PM, Bill W wrote:
However, having the proxy in the path effectively negates using IP
based
ACLS.
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Why not just set rtp-timeout-sec on the sofia profile and it'll do
that for you.
Unless something else is going on.
/b
On Dec 16, 2009, at 6:33 AM, Juan Backson wrote:
Hi,
I am having problem with around 1 % of the channels always get
zombilized.
What I want to do is to have a
see scripts/perl/call.cgi
/b
On Dec 16, 2009, at 9:59 AM, John Platts wrote:
How can I perform click-to-call or click-to-dial in FreeSWITCH?
Do you have any recommendations on programs capable of click-to-call or
click-to-dial from Microsoft Outlook or Microsoft Excel?
What SVN rev?
/b
On Dec 16, 2009, at 12:04 PM, RR wrote:
Hello All,
I know you will probably ask me to check out a fresh copy from svn trunk and
all, but I assure you I have done that yet I keep getting these errors on
make:
creating freeswitch
cc1: warnings being treated as
You have to be careful things like eyebeam will send the invite back
to FS1 that did the redirect as if it were the proxy with the request
URI as the URI you did in the 302 please post a sip trace of the
entire exchange on pastebin.
/b
On Dec 15, 2009, at 4:00 PM, Ahmed Naji wrote:
Compile it yourself is the best bet to get the very latests and
greatest code.
/b
On Dec 15, 2009, at 6:26 PM, Malay Thakershi wrote:
Is it possible to only get updated files from the latest trunk?
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I do pre releases and it'll be up shortly had to fix a couple of
bugs. I don't do binary releases for windows you'll have to do that
yourself or wait.
/b
On Dec 14, 2009, at 12:45 PM, Kendall Stauffer wrote:
HI
I tried to build the svn last Friday and it didn’t make the
sphinx dll,
Also Pre9 is up now.
/b
On Dec 14, 2009, at 1:25 PM, Jeff Lenk wrote:
Please post back to the list if you have problems with the windows
build!
Everything is working as far as I know.
If you have an existing build you should delete the following
directories
and let the scripts
if you don't have ZRTP compiled in as per the wiki it won't work...
their are a few changes coming to this code soon.
/b
On Dec 14, 2009, at 8:01 PM, Harondel J. Sibble wrote:
Hmm, I emailed the zfoneproject folks about an hour ago asking about a
release date for zfone3 and was surprised
Download MSVC and compile it yourself is usually the best bet.
/b
On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote:
Ok, So I have looked around a lot now, think I have read everything
carefully, and don’t see an answer to my questions anywhere, but
apologize if it is already somewhere.
Thats being fixed today! ;)
/b
On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote:
It hasn't been included as of late since I'm getting an unresolved
link error during the build. I'll need someone experienced in
pocketsphinx to assist with this issue:
13ngram_search.obj : error LNK2001:
Can't use G723.
/b
On Dec 11, 2009, at 5:02 AM, zendel fernandez wrote:
hi!
Pls shed some light to the below dingaling/gtalk issue.
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FreeSWITCH on windows will already poke holes in the windows firewall
using upnp. Just start FS and it works. Your outer nat is a larger
issue...
/b
On Dec 11, 2009, at 12:09 PM, Fred-145 wrote:
One last question: Does someone know of a utility for Windows that
can check
that a NAT
well mod_alas.c is for the N800 Please open a jira.
/b
On Dec 11, 2009, at 12:19 PM, Julian Lyndon-Smith wrote:
Doing the building thing, seem to have come across a bug.
Have a look at Part 2 of http://makingfs.blogspot.com/
If make crashes out, it states that it was successfully built ;)
You set the extrtp ip to an IP exactly.. this is the issue we are
fixing soon.. if you have natpmp or upnp set it to auto-nat and let it
figure it out. The issue is we have restored the behavior in 1.0.4
that lies about the IP all the time...
I'm going to commit a patch shortly that'll fix
Please test www.bkw.org/sofia_autonat_static_ip.diff
/b
On Dec 11, 2009, at 1:34 PM, Chris Chen wrote:
Thanks Mathieu, but I am on SVN r15912 now.
Chris
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You don't have to do that usually...
/b
On Dec 11, 2009, at 5:38 PM, Fred-145 wrote:
I'll see if I can find a utility that checks that the ports are open
after
FS is up and running.
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%23 is # so the question is should we URL decode that before routing?
I thought we did... what version are you using now?
/b
On Dec 11, 2009, at 5:34 PM, Michael Collins wrote:
This line is basically saying that you have a call coming from
4165551212 and it's looking for a destination
I have confirmed it works with Polycom, Snom and a few others
polycom is the hardest to set due to having to put the ca cert into
the phone... but other than that its good.
/b
On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote:
An intermediate report:
Audiocodes: TLS works only on
please look in conf/directory/default/*.xml
/b
On Dec 10, 2009, at 7:40 AM, Fred-145 wrote:
Hello
I'm going through the various XML files, and noticed this first line
in
vars.xml.
X-PRE-PROCESS cmd=set data=default_password=1234/
What is this password used for?
Thank you
Use the BKW method... three to four word sentences to describe what to
do... its very poetic! Or is that haiku?
/b
On Dec 10, 2009, at 11:53 AM, Michael Collins wrote:
I was just thinking of some way to learn FS gradually and
effectively. The
frequent problem with wiki's, is that the
fifo list issue this API and get the fifo XML and get the caller's
uuid out of the list.
/b
On Dec 9, 2009, at 10:50 AM, Luke Graybill wrote:
The short version of my question is this: how do I programmatically
determine which channel uuid the consumer channel in a fifo is
connected to?
Visit the friday meetings and we can help if you document it. ;)
/b
On Dec 9, 2009, at 3:56 PM, Tim Uckun wrote:
I found the rosetta stone useful though woefully lacking in volume.
I guess that's true overall with the project.
___
, 2009 at 11:07 AM, Brian West br...@freeswitch.org
wrote:
Visit the friday meetings and we can help if you document it. ;)
I would be willing to lend a hand with the documentation but I know so
little (a complete freeswitch noob). For example I was trying to
figure out how to tell
it out... report issues and help us make the
best FreeSWITCH release possible.
Thank you,
Brian West
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And you didn't open a Jira about this? These are the kinds of issues
that you should report so we can fix them... sitting on them and NOT
reporting them only delays the 1.0.5 release.
/b
On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote:
Changing the core db into a MySQL via ODBC caused some
I have resubmitted our request for the source.
/b
On Dec 8, 2009, at 9:58 AM, Michael Jerris wrote:
We have as of yet been unable to obtain source and we have been in
very close contact with skype all the way up to the lead technical
and business people on this project. We would of
The fun part comes when you try to link that 32bit .a file into a
64bit so file.
:P
/b
On Dec 8, 2009, at 9:49 AM, Mathieu Rene wrote:
They provide you with a 32 bit library, with the header files to
link with it.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Well the fun part is you can't link them. :P
/b
On Dec 8, 2009, at 10:38 AM, russell.mosem...@cune.org
russell.mosem...@cune.org
wrote:
That would require a dual-core processor. One core would be 32 bit and
the other core would be 64 bit. ;-)
--
Russell Mosemann
Best option for you is to use 96 in the sofia profile you're using to
talk to these broken devices.
/b
On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote:
Dear list,
Some Nec phones sends DTMF RFC2833 with payload 101 during the call,
but have negotiated a different one on SDP.
The build system for libesl and everything below that won't work 100%
on the mac just yet. You have to make some changes to how its linked
and you'll have to compile php yourself to get everything in there
properly. The perl one however is much easier to fix.
-SOLINK=-shared -Xlinker -x
session:execute(start_dtmf);
/b
On Dec 7, 2009, at 4:02 PM, Nik Middleton wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA
script? I’ve tried setting the gateway to use inband, but no joy.
It looks like I could use start_dtmf, but I can’t see how to launch
We can ONLY hope someone will do this and BSD/MIT the library and NOT
GPL it... if they GPL it then we'll have to have someone write it all
over again... love the Open Source oil and water.
/b
On Dec 7, 2009, at 7:39 PM, Jason White wrote:
it I suspect.
Given that they released the codec
Are you doing this all on a linux box thats acting as your router
too? If not you don't need two profiles... you also don't need to set
the local-network-acl on ANY profile that isn't do anything with nat.
/b
On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote:
I got a freeswitch that is
These two options attach media bugs on to the session. Which doesn't
work with native files as far as I know.
/b
On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote:
record WORKS!, but uuid_record and session_record do not want to
record in native format. do uuid_record and session_record
Is this standard recording? or voicemail?
/b
On Nov 25, 2009, at 9:11 AM, kokoska rokoska wrote:
Hello all,
is there a way how to enable very short recordings (1-3 seconds) in
FreeSWITCH other than editing source code and recompiling?
Thanks for your time!
Best regards,
You know FreeSWITCH will proxy media already if you turn off
proxy_media and disable transcoding you'll get the same results and
the IP's will be correct. Proxy media is for one purpose... T.38, it
gains you NOTHING otherwise.
/b
On Nov 25, 2009, at 10:10 AM, Juan Backson wrote:
Hi,
Yes an alias will be required for every domain you run on the profile
so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
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Or if you're dancing with the stars!!
/b
On Nov 25, 2009, at 1:55 PM, Chris Chen wrote:
One suggestion to you, please never consider the GXW4108 for any
business use unless just in LAB. The GXW4108 will work when it is
working,but I can tell you within one year you will be regretting
Kill it, sunshine.
/b
On Nov 25, 2009, at 2:40 PM, Chris Chen wrote:
You haven't really put it into production for more than one year.
The issue with GXW4108 is that after some time, say a couple of
months, either all FXO ports not working, or worse, some FXO ports
not working, but after
John,
If the remote end doesn't require a username or password then you
don't need to create a gateway to send a call to that endpoint. You
can simply do sofia/profile/num...@remoteip and it'll work.
Also can you put the patch on jira via http://jira.freeswitch.org
/b
On Nov 24,
You'll have to hairpin the media thru your machine usually if they
won't accept either of those.
/b
On Nov 24, 2009, at 3:05 PM, John Platts wrote:
How do I get FreeSWITCH to forward calls without sending 302 Moved
Temporarily or SIP REFER messages?
Are you sure you did a make current? and can you outline the issue in
more detail?
/b
On Nov 24, 2009, at 3:28 PM, John Platts wrote:
I actually checked out revision 15654 today, and I was still getting
problems with proxy media and bypass media in FreeSWITCH.
If you're doing native file you DO NOT put an extension on the file
name.
/b
On Nov 22, 2009, at 5:54 PM, Seven Du wrote:
did you try without any .wav or .PCMU?
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cd /usr/src
wget http://www.freeswitch.org/eg/Makefile
make
/b
On Nov 23, 2009, at 9:22 AM, Otis wrote:
Hi Folks
Is anyone using this on Fedora and is there a binary or installation
script anywhere
Thanks
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If you're on linux the SDK comes with the voices.
/b
On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote:
Ok. I understand that.
It would be great if someone can help me figure out:
1. Why mod_flite is not changing to the female voice even
though I tried switching all 4 profiles it
You set the ext-rtp-ip on the profile the phones talk too... but you
shouldn't be doing that.
/b
On Nov 23, 2009, at 11:08 AM, Jonas Gauffin wrote:
Hello
I got the following setup: Phones - FreeSwitch - NAT - Internet -
Gateway
And I'm struggling to get NAT working properly. I'm
outbound_caller_id is a made up variable that is used in the defaults
that are used in the examples only.
/b
On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote:
Ok. Found the problem. I had started using sofia/outbound/xxx...@sipgw2..se
as bridge destination to try to get
s/i386/x86_64/ if you are 64bit
/b
On Nov 23, 2009, at 11:47 AM, Traun Leyden wrote:
Yeah a kind user (Innotel) took the time to write up Cent OS
installation instructions for wikipbx and posted it to the wiki:
See default config it lsets you do that.
Use the variables to store two versions of the callerid then set it
depending on if its outside or inside... its rather easy to do.
/b
On Nov 23, 2009, at 11:50 AM, Jonas Gauffin wrote:
Ok. It would be a nice feature if outbound_caller_id was used
You don't have to buy the SDK... I have had it sent to everyone that
has asked me for it... the address is on the wiki for who to contact.
If you were using linux the SDK is included already.
/b
On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote:
I am not using Linux. I am using Windows
Because if you dial local-u...@local-domain thats not the correct way
this will usually trigger a call out and back in on the profile thus
moving you one leg away from the actual user. If you're going to do
that use sofia_contact and review how the defaults abstract this so
you can just
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