It's standard port 5060 UDP at both ends. I'll enable tracing on the external
profile as well - but I'm quite sure it's not used at all in this case.
I'm on my way to the lab now, so I'll soon get back with my results after using
latest SVN revision (my 'old' revision was just about 2 days old t
From your logs you need something EXACTLY like this:
http://wiki.freeswitch.org/wiki/Dialplan_XML
Everything about this topic is covered on the wiki and examples in the
default config. I don't mind helping but you do have to do a little
bit of the work you
Sorry. Could you explain more detailed?
This message means freeswitch is looking for winday in my public.xml, but it
can't find. So the dialing hang up.
I need to replace ^***, which is my DID number, to my sip account
winday?
Brian West-3 wrote:
>
> Its looking for "winday" in contex
Please refrain from posting the same question twice... I have answered
in your previous email. You might consider joining IRC and
interacting with people in realtime to answer your questions. The
channel is on irc.freenode.net and its #freeswitch, Any standard IRC
client can access the ch
Its looking for "winday" in context public so your expression would be
^winday$ Your ^***$ isn't valid as far as I can remember.
On Apr 16, 2009, at 12:06 AM, winday wrote:
Cell Phone FL->winday in context public
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.clu
most likely your hunting for 'winday' which is not found by any condition.
2009-04-15 22:57:42 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing
Cell Phone FL->winday in context public
On Thu, Apr 16, 2009 at 00:06, winday wrote:
>
> I'm a fresh user of FreeSWITCH. I just configed the b
I'm a fresh user of FreeSWITCH. I just configed the basic things. Right
now, all outbound calls are ok. I can call any number I want.
But when
I use my own mobile phone to call the phone number(I got from my sip
provider) bind with my sip account, I can not receive the call. My softphone on
my a
I'm a fresh user of FreeSWITCH. I just configed the basic things. Right now,
all outbound calls are ok. I can call any number I want. But when I use my
own mobile phone to call the phone number(I got from my sip provider) bind
with my sip account, I can not receive the call.
I configed that the
Hi,
We have several context setup with different users, all in the directory
with IP based auth.
Anyway we can turn on debug for just one context/user in the directory, when
lots of people are making calls it's hard to see traffic to just for one
user, and troubleshoot.
Also, anyway you can show
Well, it's been a few months since I mentioned this project last here,
so here's an update over my last announcement (see
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010048.html
)
Things have improved a *lot* since the last time I mentioned it:
* Support for inbound calls
Advice/Help needed...
I'm looking to use freeswitch to sbc or possibly proxy connections to my
calix ONTs.
Currently the ONTs have public ips that talk to a Metaswitch. But the
sip clients occassionally get a "sip" "sip" ring only message. They
appear to be an exploit attempt to gain the pas
What port are you hitting? Make sure you turn sip tracing on external
and internal just in case you're using either or both.
/b
On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:
I've built using latest trunk now, but I won't be able to test again
until tomorrow - I'll get back to you after
I've built using latest trunk now, but I won't be able to test again until
tomorrow - I'll get back to you after that.
Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP
Enablement Services), this one talks UDP to FreeSWITCH.
FYI, we have translation of the phrase file happening right now. But KK's
question is still valid: what does he need to do to get over the hump?
-MC
On Wed, Apr 15, 2009 at 1:27 PM, Brian West wrote:
> Have to record all the sound files in your voice at 48kHz with a good mic
> and no background
its in doc/phrase/phrase_es.xml
/b
On Apr 15, 2009, at 3:44 PM, Cesar Bermudez wrote:
can some of you provide the list of sounds or the words that need to
be recorded?
On Wed, Apr 15, 2009 at 10:27 PM, Brian West
wrote:
Have to record all the sound files in your voice at 48kHz with a
g
my voice maybe its not the best todo this, but i'have a friend that work in
a radio. ;)
On Wed, Apr 15, 2009 at 10:27 PM, Brian West wrote:
> Have to record all the sound files in your voice at 48kHz with a good mic
> and no background noise! ;)
> /b
>
> On Apr 15, 2009, at 3:13 PM, Cesar Bermu
can some of you provide the list of sounds or the words that need to be
recorded?
On Wed, Apr 15, 2009 at 10:27 PM, Brian West wrote:
> Have to record all the sound files in your voice at 48kHz with a good mic
> and no background noise! ;)
> /b
>
> On Apr 15, 2009, at 3:13 PM, Cesar Bermudez wr
Have to record all the sound files in your voice at 48kHz with a good
mic and no background noise! ;)
/b
On Apr 15, 2009, at 3:13 PM, Cesar Bermudez wrote:
what is needed to have spanish support, i'speak spanish its my
native language.
Brian West
br...@freeswitch.org
-- Meet us at ClueC
what is needed to have spanish support, i'speak spanish its my native
language.
On Wed, Apr 15, 2009 at 9:39 PM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:
> Has there been any update on this?
>
> I am willing to pay someone to add support for Spanish digits. I need
> this funct
Has there been any update on this?
I am willing to pay someone to add support for Spanish digits. I need
this functionality to finish a project.
2009/4/14 João Mesquita :
> I know spanish and I would translate it no problem. MC, get in touch with me
> off-list so we can handle that.
> I can also
I recall this and transport=tls in this case... maybe MikeJ can chime
in on this one..I thought we already fixed this.
On Apr 15, 2009, at 12:50 PM, Peter Olsson wrote:
Record-Route:
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_
My current revision is r13015. I will do an update as soon as possible and see
if that solves the issue.
Thanks!
//Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org] för Anthony Minessale
[anthony.miness..
Unfortunately this isn't being maintained and Bret didn't give his script to
any of us. If anyone out there is familiar with converting wiki pages to PDF
and is willing to pick this up then by all means contact me off list and
we'll discuss it.
-MC
On Wed, Apr 15, 2009 at 12:22 AM, Mitul Limbani
This sounds familiar:
What revision of the code is this?
Can you confirm you have this problem with SVN trunk (r13034 at the time of
this writing).
On Wed, Apr 15, 2009 at 11:24 AM, Peter Olsson <
peter.ols...@visionutveckling.se> wrote:
> This is the full SIP-trace for the call. It’s not send
This is the full SIP-trace for the call. It's not sending a BYE at all, and I
can't see one in Wireshark either. As you can see in the end there is a call to
hangup_function(), but no SIP messages after that. When I manually hangup the
phone I can see it sends BYE to FreeSWITCH (which is quite e
Hi,
I have two scenarios I'm having trouble figuring out and I'd be happy if
someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a forked dial with
the following:
However the second leg is called at exactly the
type: sofia profile internal siptrace on at the cli and try again
see if you cen see FS sending BYE to the wrong address.
This can be caused by a false positive on the NAT detection or when you need
NAT mode and you don't have it enabled.
first edit the sofia profile in your config and comment o
Press F8 and try again... With debug cranked up you'll see more details.
On Apr 15, 2009, at 10:10 AM, Antony King wrote:
sumably there's some difference between calls coming in via a
gateway and localy generated calls; could someone give me some
pointers as to how to get it to accept the ca
I'm just getting started with freeswitch; I'd like to create a public phone
number that a small number of people can
dial into to join a conference.
I've got calls and the conference rooms working internally, and I've got a link
to my sipgate account which directs to
extension 1000 . The confi
Has anybody tried this site?:
- http://www.pcapr.net/browse/voip
Looks pretty good for posting sip dumps and the like. It's a bit more
sophisticated than pastebin.
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.free
You have to determine how far it will scale for YOUR needs nobody can
answer this question. It all depends on what YOU are doing with it
and how crazy wild you go with things in your implementation. ;)
/b
On Apr 15, 2009, at 8:59 AM, Parveen Kumar Jain wrote:
Hi,
I need to develop a
Chris,
I am able to reproduce this issue so now you'll need to open a jira
on this issue please follow the guidelines here http://wiki.freeswitch.org/wiki/Reporting_Bugs
On another note PLEASE do not hijack threads. You clicked reply..
cleared the body and the subject and sent the message.
> Hi,
>
>I need to develop an IVR application which makes an outbound calls and
> then plays the some audio file for the user. For this I was trying to
> evaluate Freeswitch under following criteria:
>
> - Does freeswitch have outbound calls support(is there any conf file file
> avilabel where
Can you describe the call path a bit more and what SVN rev are you on?
/b
On Apr 15, 2009, at 8:43 AM, Peter Olsson wrote:
When I do a call from my Avaya SIP Server to FreeSWITCH. And then
let FreeSWITCH do a hangup of the call, FreeSWITCH doesn’t seem to
send a ”BYE” back to the Avaya PBX.
FreeSWITCH Version 1.0.trunk (12933M)
Has anyone else run into this issue? Here is what I'm doing.
2 Callers call in (both can be internal; using a snom m3 and a snom 370
in this test).
Case 1:
When both callers dial in they are directly routed into a conference
room via a dialplan "myConfere
When I do a call from my Avaya SIP Server to FreeSWITCH. And then let
FreeSWITCH do a hangup of the call, FreeSWITCH doesn't seem to send a "BYE"
back to the Avaya PBX. I've narrowed it down to this simple example in the
dialplan;
In this case no BYE is issued, and the pho
Baskar pisze:
> *Hi,
>
>Now i can able to load the mod_java in the freeswitch console.
>
> After that i have followed these method to run the PhoneTest.java
>
> *
>
> *1) verified my classpath in the java.conf.xml: value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar"/>*
*Hi,
Now i can able to load the mod_java in the freeswitch console.
After that i have followed these method to run the PhoneTest.java
*
*1) verified my classpath in the java.conf.xml: *
*2)my PhoneTest.class is located in /usr/local/freeswitch/script -
directory, same as where freeswitch.j
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