will register with n1.domain.com or
n2.domain.com
Thats how it matches it up.
/b
On Mar 26, 2008, at 12:28 PM, Ivan C Myrvold wrote:
I am not sure if this is a bug, or if this is how it should behave:
If you in conf/directory have two files, say n1.xml and n2.xml, and
two directories n1
What about an Installer using PackageMaker for OS X?
Ivan
Den 21. mai. 2008 kl. 20:29 skrev Michael Jerris:
I have done a very basic msi that visual studio builds, but it does
not include bootstrapper for the runtime or the sound files. If
someone has good experience with doing a more full
On May 21, 2008, at 3:12 PM, Ivan C Myrvold wrote:
What about an Installer using PackageMaker for OS X?
Ivan
Den 21. mai. 2008 kl. 20:29 skrev Michael Jerris:
I have done a very basic msi that visual studio builds, but it does
not include bootstrapper for the runtime or the sound files
at the Asterisk to FreeSWITCH section on the wiki.
http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk
/b
On Jun 17, 2008, at 12:31 AM, Ivan C Myrvold wrote:
I have used Freeswitch with a DID from Voxbone working on IAX, and
that have been working so well. But now Voxbone
configure FreeSwitch to allow the call into FreeSwitch
without proxy authentication?
Ivan
Den 18. juni. 2008 kl. 09:25 skrev Ivan C Myrvold:
I added to my acl.conf.xml this
http://pastebin.freeswitch.org/4640
and this to external.xml
param name=apply-inbound-acl value=voxbone/
When I
to the list and see if that makes a
difference.
/b
On Jun 20, 2008, at 11:28 AM, Ivan C Myrvold wrote:
I do not have outbound registation to Voxbone, because Voxbone is
only incoming. I am not registrating Voxbone at all.
In their FAQ, they have how to configure for Asterisk
, which is not forwarding RTP ports
correctly, or is there still a piece in FreeSwitch I have missed?
Ivan
Den 20. juni. 2008 kl. 22:52 skrev Brian West:
Good to know it snapped into place now! :P
/b
On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote:
This is all making sense to me now
I put this into internal.xml, and this seems to do the trick:
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
Ivan
Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold:
Yeah, the call arrives nicely now, but the audio is only 1-way.
I
My FreeSwitch box is sending these OPTIONS sip messages to my Cisco
7960 phone at 192.168.207.160. What does it mean?
It is sending these messages even when I unplug the Cisco phone:
http://pastebin.freeswitch.org/4669
Ivan
___
Freeswitch-users
In my dialplan, I have set the following to send incoming calls to an
application I am working on:
action application=socket data=192.168.207.242:8084 async full/
This is working very nice, and I am receiving events for all incoming
calls, and I am redirecting the calls in my application to
FreeSWITCH Version 1.0.pre4 (8805)
Ivan
Den 22. juni. 2008 kl. 14:40 skrev Brian West:
What rev are you using?
/b
On Jun 22, 2008, at 6:07 AM, Ivan C Myrvold wrote:
My FreeSwitch box is sending these OPTIONS sip messages to my Cisco
7960 phone at 192.168.207.160. What does it mean
I do a sendmsg and execute a bridge sofia/
192.168.207.203/116%192.168.207.203
Ivan
Den 22. juni. 2008 kl. 14:40 skrev Brian West:
What are you doing in your script over the socket? You're directing
them elsewhere?
/b
On Jun 22, 2008, at 7:36 AM, Ivan C Myrvold wrote:
In my dialplan
SendMsg
call-command: execute
execute-app-arg: sofia/192.168.207.203/116%192.168.207.203
execute-app-name: bridge
Ivan
Den 22. juni. 2008 kl. 14:40 skrev Brian West:
What are you doing in your script over the socket? You're directing
them elsewhere?
/b
On Jun 22, 2008, at 7:36 AM, Ivan C
I see in the documentation that I should send a Sendmsg uuid, and
not a SendMsg without the uuid. Maybe that will give me all messages
with this uuid?
Ivan
Den 22. juni. 2008 kl. 14:59 skrev Ivan C Myrvold:
SendMsg
call-command: execute
execute-app-arg: sofia/192.168.207.203/116
the socket for the terminated call close;
the others stay open.
From: Ivan C Myrvold
To: freeswitch-users@lists.freeswitch.org
CC: Greg White
Sent: 06/22/2008 09:39 AM
Received: 06/22/2008 09:39 AM
Subject: Re: [Freeswitch-users] Outbound socket
That doesn't help if you have several calls going
I got email from support today, so hopefully they will find out why
this happens, although they asked me to try out some other ports,
which I did and the same happened.
Ivan
Den 24. juni. 2008 kl. 11:10 skrev Ivan C Myrvold:
I have a softphone, iSoftPhone, I am trying out, but when
I had a conference in FreeSWITCH with a few of my friends, and one of
my friends complained that when I talked, at every pause I had in my
talk, the first word in my sentence was cut off.
When no one is talking, there is completely slience.
Is this behaviour configurable? I saw something in
My FreeSWITCH is behind nat, and I have calls coming in from internet
on port 5060. This works great after I got the IP range set in
acl.conf.xml.
But the calls get hung up after 40-50 seconds into the call for no
reason at all. But after I restart FreeSWITCH, this annoying behaviour
I should add that after 1 day, the same happens again. So I have to
restart FreeSWITCH at least once a day.
Ivan
Den 9. juli. 2008 kl. 23:16 skrev Ivan C Myrvold:
My FreeSWITCH is behind nat, and I have calls coming in from
internet on port 5060. This works great after I got the IP range
).
Ivan
Den 9. juli. 2008 kl. 23:39 skrev Brian West:
Can you update to the latest your may rev's behind?
On Jul 9, 2008, at 4:29 PM, Ivan C Myrvold wrote:
I should add that after 1 day, the same happens again. So I have to
restart FreeSWITCH at least once a day.
Ivan
Den 9. juli. 2008 kl. 23
them as when you have the options and you don't need them so
try both.
On Fri, Jul 11, 2008 at 1:30 PM, Ivan C Myrvold [EMAIL PROTECTED]
wrote:
I updated to latest svn, but exactly the same is happening now. This
is a SIP trace from before I restart FreeSWITCH:
http://pastebin.freeswitch.org
.
/b
On Jul 12, 2008, at 8:44 AM, Ivan C Myrvold wrote:
Yes, of course, else it would not have worked at all.
My point is, it looks like that only some of the IP addresses in the
range work, for some reason.
Ivan
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Freeswitch-users mailing list
http://jira.freeswitch.org/browse/MODAPP-113
Ivan
Den 12. juli. 2008 kl. 16:52 skrev Brian West:
Can you open a bug on jira please... this isn't working correctly.
I just verified. I might be wrong but lets open a jira and make sure.
/b
On Jul 12, 2008, at 9:24 AM, Ivan C Myrvold wrote
I have
FreeSwitch Version 1.0.trunk (9031)
and have no such problem.
Ivan
Den 16. juli. 2008 kl. 13:17 skrev UV:
In recent builds the event socket module won't allow authentication.
Any ideas why?
Was tested on build 1.0.9009M both on Windows and CentOS.
Here's what you get (on telnet
I have a NAT problem with my FreeSWITCH which shows up only when my
public address have changed AFTER I have started FreeSWITCH.
I don't know if this is a small bug in FreeSWITCH, therefore I want to
get your opinion before I post a Jira ticket:
I have an incoming SIP call from a registered
I am testing out conditions in my dialplan, but am unable to get a
match on caller_id_number, and I wonder why.
I have to numbers registered from my PAP2, with User ID 100, and 200.
If I have this in my dialplan, I thought I should get a match if I
call any number from any of these two
:
try ${caller_id_number}
/b
On Aug 15, 2008, at 3:58 PM, Ivan C Myrvold wrote:
caller_id_number
Brian West
sip:[EMAIL PROTECTED]
___
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman
in the expression both caller_id_number and ${caller_id_number}
will work if you do not nest conditions.
you do not say ${variable_caller_id_number}
On Fri, Aug 15, 2008 at 3:58 PM, Ivan C Myrvold [EMAIL PROTECTED]
wrote:
I am testing out conditions in my dialplan, but am unable to get a
match
%imyrvold.dyndns.org/
/condition
/extension
Ivan
On 16. aug.. 2008, at 18:59, Ivan C Myrvold wrote:
OK, no nested conditions. But I am still unsure about the syntax. In
the wiki page I could not find any example of stacked condition.
I tried with the below syntax, but still no success
=effective_caller_id_name=Ivan C
Myrvold/
action application=set
data=effective_caller_id_number=37048300/
action application=bridge
data=sofia/gateway/ip24_37048300/$1/
/condition
condition field=caller_id_number expression=^140$ break=on-
true
action
application=set data=dialed_ext=$1/
/condition
condition field=caller_id_number expression=^100$ break=on-
true
action application=set data=effective_caller_id_name=Ivan C
Myrvold/
action application=set
data=effective_caller_id_number=37048300/
action application=bridge
Sorry about that. Yes, you are absolutely right. Now it works perfect.
Thanks again!
Ivan
On 17. aug.. 2008, at 11:07, Brian West wrote:
You didn't pay attention to what I did. Notice in your first
condition I set dialed_ext=$1
/b
On Aug 17, 2008, at 4:00 AM, Ivan C Myrvold wrote:
Yes
, 2008, at 10:11 AM, Ivan C Myrvold wrote:
I thought I would register my telephone number with e164.org, so that
anyone can call my number with SIP.
I am not going to have to make a new profile for this, as the calls
will come in on port 5060, right?
I see the test call coming in on my FreeSWITCH
are kidding. Then I found out its working too.
Thats coool.
On Fri, Aug 29, 2008 at 7:51 PM, Ivan C Myrvold [EMAIL PROTECTED]
wrote:
or
...
(i.e. three periods).
Ivan
Den 29. aug.. 2008 kl. 12:54 skrev Adeel Ansari:
shutdown
On Fri, Aug 29, 2008 at 6:49 PM, Sunil Singh
[EMAIL PROTECTED
directly in my OS (Mac OS X). It
sounds exactly like the .wav file.
But with the help of what program/filter? Plaing directly - is
made by means of any OS's programs too.
We works under CentOS 5.1
From: Ivan C Myrvold
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, September 09, 2008 1
Yes, you can. Check out event_socket.conf.xml
Ivan
Den 10. sep.. 2008 kl. 08:24 skrev Jonas Gauffin:
Hello
I'm running FS as a NT service. Is it possible to access the FS
console through telnet or something like that?
//Jonas
___
Is FreeSWITCH on a public IP, or behind nat?
Ivan
Den 16. sep.. 2008 kl. 13:44 skrev Jonas Gauffin:
Hello
I haven't touched my sofia profiles in a long time.
I now need to have two profiles for phones registering against
FreeSWITCH:
* One for phones that are on the same LAN as FS
*
Remove the second ^ and see if that helps.
Ivan
Den 17. sep.. 2008 kl. 12:52 skrev xbipin:
with the following expression i cant dial any sipbroker number, eg:
*01118
^(^\*\d+)$
--
View this message in context:
http://www.nabble.com/regexp-help-tp19529525p19529525.html
Sent from
I have for more than 1 year now sent my inbound calls to an
application running on another machine listening to FreeSWITCH's
outbound socket.
I have this in my dialplan:
action application=socket data=192.168.207.101:8084
async full/
In my external application, I look
Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it
out, and it works perfect with my FreeSWITCH installation.
Can be downloaded at iTunes App Store.
Ivan
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
On 4. okt.. 2008, at 14:14, Rupa Schomaker wrote:
On 10/4/2008 4:47 AM, Ivan C Myrvold wrote:
Finally an iPhone app that connects with SIP to FreeSWITCH. I tried
it
out, and it works perfect with my FreeSWITCH installation.
Can be downloaded at iTunes App Store.
Ivan
Ivan, I've been
Den 17. okt.. 2008 kl. 14:45 skrev Dennis:
One of the problems I have with outbound is, that I won't get all
events. For example I do not get hangup, because the socket
connection seems to die, before freeswitch sends the last command when
hanging up. With ringing the problem seems to be the
Did you read carefully when asked to provide login and password?
The login and password is there, don't use your own freeswitch login.
Ivan
Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain [EMAIL PROTECTED] wrote:
Hi,
I'm writing an IVR in Lua and am
I found out that both /event and /events worked as commands, but
only /noevents worked, not /noevent, although the Wiki says /
noevent.
Ivan
Den 30. des.. 2008 kl. 08:24 skrev Michael Collins:
Ken,
Thanks for the clarification. I will make a note of this in the
wiki. Also, can you hum a
I haven't tried using launchctl for FreeSWITCH. But when I saw your
post, I tried it out. I have no problem getting it to work:
I make a file org.freeswitch.freeswitch.plist and save it to ~/
Library/LaunchAgents with the following content:
?xml version=1.0 encoding=UTF-8?
!DOCTYPE plist
it in /Library/LaunchDaemons if you run
it as root (but you should in my opinion run it as a normal user, as I
have (almost) always done).
Ivan
Den 16. jan.. 2009 kl. 20:12 skrev Martin Joseph:
On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote:
I haven't tried using launchctl for FreeSWITCH
Den 17. jan.. 2009 kl. 19:15 skrev Martin Joseph:
On Jan 16, 2009, at 12:54 PM, Ivan C Myrvold wrote:
I have chown the freeswitch directory to my user imyrvold, therefore
I put it in ~/Library/LaunchDaemons.
Do you run freeswitch as root, as you put it in /System/library/
LaunchDaemons
s it possible to run Skypiax on OS X? The wiki says Linux and Windows,
but says nothing about OS X.
I have been running FreeSWITCH on OS X for a couple of years now, and
love it. Adding Skype gateway would be really sweet.
Are there any plans for adding Skypiax to trunk, or do we have to
Is skypiax now working on Mac OS X in Freeswitch?
Ivan
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
:
I'm not sure about that one I haven't tried lately because the
API
differs on the Mac last I looked at it.
/b
On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
Is skypiax now working on Mac OS X in Freeswitch?
Ivan
___
FreeSWITCH-users
I haven't tried lately because the
API
differs on the Mac last I looked at it.
/b
On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
Is skypiax now working on Mac OS X in Freeswitch?
Ivan
___
FreeSWITCH-users mailing list
FreeSWITCH-users
On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
Is skypiax now working on Mac OS X in Freeswitch?
Ivan
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
if it helps. I bet you can make it work. Also code will be
in my branch soon.
7.
On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
Yes, I am interested in this, and if you have any source I could have
a look at it.
Ivan
Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
On Aug 9, 2009, at 11
I have got outgoing call to Skype to work, and the audio quality is
excellent. But I also have problems with incoming call, looks like it
doesn't get bridged by FreeSWITCH. I put sofia on debug, but couldn't
see any Sofia messages at all. Only message I saw in the console was
this:
, at 3:19 PM, Ivan C Myrvold wrote:
I have got outgoing call to Skype to work, and the audio quality is
excellent. But I also have problems with incoming call, looks like
it doesn't get bridged by FreeSWITCH. I put sofia on debug, but
couldn't see any Sofia messages at all. Only message I saw
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had no audio, in neither
direction. Incoming
:46 AM, Ivan C Myrvold i...@myrvold.org
wrote:
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had
you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia SIP profiles involved.
On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold i...@myrvold.org
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org
wrote:
Here is a debug log from a call from an internal phone out
the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.
On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold i...@myrvold.org
wrote:
Oh, what happened to it?
Anyway
siptrace on
sofia loglevel all 9
^
Then run your call, then do this:
sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning
On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold i
. 2009, at 21:26, Ivan C Myrvold wrote:
Yes, now I got a more detailed trace. Thank you for helping me with
this.
A new pastebin at http://pastebin.freeswitch.org/10905
Ivan
Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:
fsctl loglevel debug
console loglevel debug
sofia profile internal
: 200 OK has fatal syntax errors
This is a know-bug in asterisk.
see: https://issues.asterisk.org/view.php?id=15621
On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold i...@myrvold.org
wrote:
No one have any idea why this is not working? I have combed through
the log, but couldn't find any clue
FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my
intel Macs with great success.
I am also developing a GUI application using Cocoa. I started that a year ago,
but haven't looked at it for a while, but this Christmas I have started working
on it again.
Ivan
Den 27.
On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote:
FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my
intel Macs with great success.
I am also developing a GUI application using Cocoa. I started that a year
ago, but haven't looked at it for a while, but this Christmas I
I am using iSoftPhone, works great with FreeSWITCH.
Ivan
Den 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
, 2009, at 3:35 PM, Ivan C Myrvold wrote:
I am using iSoftPhone, works great with FreeSWITCH.
Ivan
Den 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
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