Hi,
I want establish a trunk between avaya communication manager to
freeswitch (h323/Sip trunk ).
Is any body trying to establish these trunk ..?
Please forward the details with configuration settings from avaya side and
freeswitch side.
Regards
Samba
Mike,
Looks like upgrading to trunk ... let me know if you have any thoughts.
Here is the trace:
U 4.71.122.167:5060 - 4.71.122.130:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 4.71.122.130;branch=z9hG4bK340a.42319c57.0.
Via: SIP/2.0/UDP
Hi, I am getting 406, no matter what. I have tried 3 different INVITEs.
--
*INVITE sip:[EMAIL PROTECTED] [EMAIL PROTECTED];transport=udp
SIP/2.0
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
From: sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=5919
To: sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
Via:
This isn't strange at all its part of the default configs.
/b
On Nov 11, 2008, at 9:28 AM, [EMAIL PROTECTED] wrote:
What does your dialplan look like? I see the error, but I can't quite
tell what is wrong. It looks like there is some sort of strange
variable assignment going on variable
That isn't the problem because I run with this config every day in my
testing.
/b
On Nov 10, 2008, at 11:50 AM, Peter P GMX wrote:
Addendum:
I fixed the sip_secure_media=[undef] problem. However no change.
Best regards
peter
___
But what does this error message mean? I started with the latest SVN
version with standard setup, tried to do an internal call (all phones
are behind NAT on different public IPs) and it failed.
I cannot reach either of the phone, although sofia status profile
internal shows me the correct
Try updating to latest trunk and you will probably see user_not_registered
instead.
If you already are on latest it means you are asking for an invalid sofia
profile.
On Tue, Nov 11, 2008 at 9:35 AM, Brian West [EMAIL PROTECTED] wrote:
This isn't strange at all its part of the default
Thanks Anthony! That seems to have done the trick.
SR
On Tue, Nov 11, 2008 at 8:10 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
you can work around this issue by disabling 100rel
param name=enable-100rel value=false/
in the sip profile.
We have an issue open with sofia to correct this?
My patch was added to the trunk weeks ago. Download the latest version of
mod_fifo and you will see it.
The event fifo::info with action=consumer_reentrance is fired just as the
consumer leaves wrapup iff he entered the fifo in wait mode and has not hung
his call.
It is around line 133x
Bye.
Hello
Do anyone know where I can find a xml/csv file with international/national
dialing codes?
Thanks,
Jonas
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Nobody has a clue why I cannot connect the 2 phones internally? I
googled around, I can see some other people asking for this problem at
various places, but no reply so far.
Best regards
Peter
Peter P GMX schrieb:
Addendum:
I fixed the sip_secure_media=[undef] problem. However no change.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I want a dialplan setting a channel variable with a certain value. After
that I want to do some action depending of that variable/value. But it
looks like that FS parses the regular expressions of dialplan conditions
only at the beginning of
I did an install via make current and thus updated to revision 10325. It
built completely and all modules and apps have the current date/time.
I still have the old configs from revision 10306 though.
mod_dptools is loaded and concerning mod_dptools I get in the logs:
2008-11-11 19:46:23 [DEBUG]
is that all of the logs or are you truncating it.
start FS like this
TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch
and capture 100% of the console log
and paste it to
http://pastebin.freeswitch.org
On Tue, Nov 11, 2008 at 12:56 PM, Peter P GMX [EMAIL PROTECTED] wrote:
I did an
I would like to know if anyone is familiar with the Freeswitch proxy
package, Liverpie(http://www.liverpie.com)? If so, what have your
experiences been with this package?
Jonathan
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
1. you need to include an SDP or enable 3pcc support on the sip profile...
2. the 3rd invite looks like it doesn¹t include any codecs (or atleast any I
am familiar with)
From: Adeel Ansari [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 11 Nov 2008 17:29:50 +0800
Hi Guys,
I have this problem when i'm trying to reproduce more than 8 prompts in a
row.
Here is code extract that simplifies the problem i'm having.
If put something like this inside a script.py
session.execute(playback,1.wav)
session.execute(playback,2.wav)
we read every single reply and we make socket_read with the legth
returned by Content-Length.
what we can do with fs, socket outbound and our php-script:
1.) we can answer the inbound, make a bridge to another phone and both
lines are connected and can talk to each other.
2.) we can make an
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Birgit,
well of course the condition is THE condition, but I espected the whole
dialplan is executed sequentially, so that this fictive dialplan works:
condition field=xyz expression.* break=never
action application=set data=MYVAR=abc/
Actually, I am just trying to build a minimal client/UA for Freeswitch. Do I
need to go into 3pcc or something. I remember I did play with 3pcc but at
that time I was implementing a sip stack and trying to come up with a
minimal softswitch. Am I misunderstanding something?
It sounds like, I need
Hi Helmut,
As far as I know, the condition at the top of the dialplan is indeed
THE condition.
We've used two mechanism for 'dynamic' dialplan:
- xml_curl: freeswitch posts parameters to a URL and the URL returns a
dynamic dialplan. This works well with a webserver connected to a database
-
If you look very close in the dialog box it says what they are. Its
pastebin and freeswitch
You failed the test :P
/b
On Nov 11, 2008, at 4:14 PM, Peter P GMX wrote:
http://pastebin.freeswitch.org http://pastebin.freeswitch.orgasked
for
login credentials. Any idea where to get them from?
Aaargh, being able to read can be a real advantage sometimes.
I have now put the log to
http://pastebin.freeswitch.org/6098
Best regards
Peter
Brian West schrieb:
If you look very close in the dialog box it says what they are. Its
pastebin and freeswitch
You failed the test :P
/b
On
The second part is in:
http://pastebin.freeswitch.org/6099
Peter P GMX schrieb:
Aaargh, being able to read can be a real advantage sometimes.
I have now put the log to
http://pastebin.freeswitch.org/6098
Best regards
Peter
Brian West schrieb:
If you look very close in the dialog box
It appears to have been cutoff. The last line that I see is:
2008-11-11 23:38:27 [NOTICE] switch_loadable_module.c:281
switch_loadable_module_process() Adding F
Peter P GMX wrote:
Aaargh, being able to read can be a real advantage sometimes.
I have now put the log to
FYI,
There is like a 1400 line limit in pastebin, or something like that, so
you have to be careful when you have a large dump to put there.
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, November 11,
I have tried to raise this... so I have a request here for someone in
the community to step up and either write us a new pastebin that
doesn't suck or find me one that doesn't take 100 depends to install.
So if you wish to help with the pastebin please email me off list.
/b
On Nov 11,
it means your user is not registered.
you are calling
user/[EMAIL PROTECTED]
and there is no user registered with that [EMAIL PROTECTED]
On Tue, Nov 11, 2008 at 4:51 PM, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
It appears to have been cutoff. The last line that I see is:
2008-11-11
regarding:
it means your user is not registered.
Before I made the call I registered both phones manually (btw. I can
make calls from both phones).
Then I did a sofia status profile internal and it shows me both phones
registered on that domain:
1.
API CALL [sofia(status profile
what is the issue chris
On Tue, Nov 11, 2008 at 5:52 AM, Cristian Talle [EMAIL PROTECTED] wrote:
Hi,
I'm having some issues retrieving a dtmf sequence using getDigits.
The xml dialplan invokes a javascript file that in turn runs:
var digits = ;
session.answer();
digits =
Hi,
Here is the exact lua script:
digits = session:playAndGetDigits( 1, 1, 3, 3000, #,
phrase:enter_userid,phrase:invalid_input,[1|2|3|4|5|6])
If I repeatedly enter 9, here is what would happen:
z
2008-11-12 22:48:22 [DEBUG] switch_ivr_play_say.c:269
switch_ivr_phrase_macro() Handle
Hi,
*Just i want to call the 1002==9841799874 and i want to call another no
9840544078 and add in to the conference room *
*First method i tried :*
api originate sofia/internal/1002%172.20.191.227 bridge(sofia/default/
[EMAIL PROTECTED]) 1002 call 9841799874
api uuid_transfer
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