Hi,
If I have different audio tracks that I would like to loop through, does it
mean I need different folder for each? Moreover, if I loop through
different folders, does it mean freeswitch would start multiple "player" and
consume resources ineffectively as the audio is probably played only for
Hi
> If you use a sofia profile with ODBC on multiple machines, if you register
> to any machine all of them will resolve the correct contact address when
> trying to make an outbound call. Presence, however such as subscriptions
> must still go to the original box because there is an open SIP d
Hi,
I am hoping to get some deployment insights from people who have implement
freeswitch on a large-scale. I have followed the wiki in setting up Xen and
Ultramonkey to load balance two freeswitch. But since freeswitch is
stateful and two freeswitch can't share info such as fifo, I am wondering
Hi,
Have you tried the acl function in freeswitch. For example, setup the DIDWW
domain as an acl so no need to use username/password from that domain.
In the default profile, you can specify the default context for that
incoming call to go to in the dialplan.
If my solution works for you, please
Hi,
I am trying to get events to be fired out using the following codes in JS:
e = new Event("custom", "message");
var txt = "main menu===";
e.addBody(txt);
session.sendEvent(e);
Within my client, I sent "Plain event all" to listen to all events.
Then, I filtered on all the events, but
Hi,
I am unable to get FS to send and receive faxes and I hope someone can give
me a hand.
When sending the fax, I get the following error from sock2me:
Sending Fax filename: [/home/anne/abcd.tiff] from 127.0.0.1:9015 ->
127.0.0.1:9016
Phase B handler on session 0 - (0x80) DIS
Phase E handler on
Hi,
I have tried out the following command from Telnet:
api originate sofia/default/1005 1 &park
Content-Type: api/response
Content-Length: 41
+OK 8dc747d4-1261-11dd-92a8-2750a3aa164c
SendMsg 8dc747d4-1261-11dd-92a8-2750a3aa164c
Call-command: execute
execute-app-name: fifo
execute-app-arg:
idea?
Thanks,
Pete
On Wed, Apr 23, 2008 at 9:15 PM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> your domain and user name are both blank in your xml
>
> On Wed, Apr 23, 2008 at 5:10 AM, Pete Kay <[EMAIL PROTECTED]> wrote:
>
> > Hi,
> > I am having a voic
Hi,
I am having a voicemail problem:
Problem 1
In the wiki, it says that with the below line, the user can check vm without
needing to authenticate:
Instead, it goes to the mailbox asking user to record message:
[/usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-record_messa
Hi,
I am in the processing of implementing a typical scenario for service call.
So, in my scenario, I would have several agents who may log in and log out
of the queue at different times of the day.
People can call in and will be routed to one of the available agents. If no
agent responses within
Hi,
I have a need to extra a list of valid DTMF input from DB ( as it may
sometimes chang). The way I would do it is to extract the list of valid
DTMF from DB before session.answer(). When the caller enters the DTMF when
the menu is played, I need to search to see if the input is within the the
Hi,
I have a question about TTS efficiency. I have some text that I need to
read out when someone calls in and these texts do change. Here are two
options I am thinking about:
In my configuration, I would have DB and FS on different servers. DB server
is SCSI.
Option 1:Store the text in DB
Wi
Hi,
I am wondering if storing and retrieving voicemaessage / greeting from DB
makes sense especially for a load balancing environment. Is there anyway to
do it in freeswitch? If not, is there any way of using some ftp-style
storing and retrieving?
How is it done typically to have voicemail work
10:25 PM, Brian West <[EMAIL PROTECTED]> wrote:
>
> On Apr 20, 2008, at 8:54 AM, Pete Kay wrote:
>
> > Hi,
> >
> > I two a few questions about hangup and sounds:
> >
> >
> > 1.
> > In my dialplan, I have
> >
> >
> >
&g
Hi,
I two a few questions about hangup and sounds:
1.
In my dialplan, I have
After it play the call-back-later marco, it then goes on to play the "The
extension you are dialing is unavailable..." even I have specified the
HANGUP application.
Hi,
There are a few problems I found with voicemail and I am not sure if I did
something wrong:
Problem 1:
With is my dialplan:
This is the result:
2008-04-19 22:12:35 [DEBUG] switch_ivr_play_say.c:115
switch_ivr_phrase_macro() No language specified - Using [zh]
2008-04-19 22:
he etpan? Will the voicemail.js work without it?
Is etpan needed and how to get it to work? I can't find the module etpan.
Thanks alot.
Pete
On Fri, Apr 18, 2008 at 10:05 PM, Brian West <[EMAIL PROTECTED]> wrote:
> This is a huge clue.. check permissions.
> /b
>
> On Ap
Hi,
I am having problem with the speak marco, can someone please help me?
In the sample code:
session.sayPhrase("speak", "Please leave a message after the beep", "en"
);
Inside /usr/local/freeswitch/conf/lang/en/vm/sounds.xm, I added:
I am getting
Hi,
I am encountering a number of problems when trying to run the voicemail.js
provided as one of the samples:
1. the etpan module is not available and I can't find the etpan module in
the module.conf, so I have to comment out the below line.
use("etpan");
So, what is the usage of the etpan?
Hi,
It seems like the reloadxml function in fs still requires the server to
restart to take effect of new update. Is it the case or it is just my set
up problem?
Sorry to ask again. I am getting a bit frustrated. I can login now, but
the user_context info does not get set. The following user
Hi,
>From the debug file created by fs , I can see the response as follows:
But it still does not authenticate my user. Why? I have checked the format
which is same as the one in the wiki doc.
Thanks,
Pete
_
Hi,
I have a questions about multi-language voice support in FS. How to I add a
new module such as mod_fr.so? I can make the necessary wav files, but how
to create the modules?
Thank you very much for your inputs.
Regards
Pete
___
Freeswitch-users m
am suspecting that the info pass is not
sufficient for generating the directory xml. Is there anyway to see the
HTTP request and response from the console? I turned on curl_xml debug_on
but still it does not show.
Any inputs will be greatly appreciated.
Thanks,
Pete
On Fri, Apr 18, 2008 at 2:28 PM
Hi,
I place my xml response generator directory.php under
/var/www/fs/directory.php
In my xml_curl_conf.xml, I have
http://localhost/fs/";
bindings="directory.php"/>
2008-04-18 22:23:17 [ERR] mod_xml_curl.c:251 do_config() Binding has no url!
2008-04-18 22:23:17 [NOTICE] mod_xml
ED]> wrote:
> Let me guess you're running FS and X-Lite on the same machine? What was
> your problem again? From the looks of this you have changed the default
> configs a little bit.
>
> /b
>
> On Apr 17, 2008, at 11:48 AM, Pete Kay wrote:
>
> Hi Brian,
>
&
Hi Brian,
Here is the full log. Please kindly take a look:
2008-04-18 08:51:44 [DEBUG] switch_core_session.c:730
switch_core_session_thread() Session 14 (sofia/default/
[EMAIL PROTECTED]:5061) Locked, Waiting on external entities
2008-04-18 08:51:44 [NOTICE] switch_core_session.c:748
switch_core
Hi,
I comment the line at default.xml
But, it still does not work.
2008-04-18 07:38:32 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1002->[EMAIL PROTECTED]
2008-04-18 07:38:32 [INFO] switch_ivr_async.c:1357
switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::
ttp://wiki.freeswitch.org/wiki/Sofia )
> --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Pete Kay
> *Sent:* Friday, April 18, 2008 1:08 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users]
the "Local_Extension" section is says:
> >
> > Where it should actually be:
> >
> >
> > Check
> http://wiki.freeswitch.org/wiki/Sofia#Call_a_locally_registered_endpoint
> >
> > Cheers,
> > UV
> > From: [EMAIL PROTECTED] [mailto:
> [
Hi,
When I do reloadxml, I keep on getting this error:
[EMAIL PROTECTED]> reloadxml
Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml
API CALL [reloadxml()] output:
Is this another bug in the sample files? If so , how do i fix it?
Thanks,
Pete
__
lease help me so I get get started with FS.
Thanks,
Pete
On Thu, Apr 17, 2008 at 10:48 PM, UV <[EMAIL PROTECTED]> wrote:
> Try this instead:
>
>
>
>
> --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *P
Hi,
I am still not yet able to get one sip phone to call the other due to the
problem I posted. So, I used the working sip client to try to call an
outside number
by setting up a gateway.
I tried to set up fastswitch to route call to my voip provider, I am getting
channel error but can't know wh
Hi,
I am working on some test on seeing how I can port my exist Asterisk stuff
to Freeswitch. I am just getting started and I am hoping someone can give
me some help to get started.
I installed with all the default config and xml setting. Then, I bring up
two SiP clients - one in the same mach
Hi,
I made a silly mistake. The problem was solved. Please ignore.
Thanks.
Regards,
Pete
On Thu, Apr 17, 2008 at 12:38 PM, Pete Kay <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I followed the online installation but when I executed, make sure or
> make moh, I am getting th
Hi,
I followed the online installation but when I executed, make sure or
make moh, I am getting the following errors:
ser:/usr/src/freeswitch-1.0.rc2/build# make moh-install
make: *** No rule to make target `moh-install'. Stop.
ser:/usr/src/freeswitch-1.0.rc2/build#
What could be the problem?
Hi,
Two more questions about Freeswitch. I have some application written
to send and receive fax using Hylafax and IAXModem, will I be able to
port that to Freeswitch?
What is the best way to connect Freeswitch to an E1? Does anyone has
experience with any easy-to-use, easy-to-maintain hardware
-time does?
Thanks alot for your inputs.
Regards,
Pete
On Thu, Apr 17, 2008 at 12:48 AM, Brian West <[EMAIL PROTECTED]> wrote:
>
> On Apr 16, 2008, at 11:44 AM, Pete Kay wrote:
>
> > Hi,
> >
> > I have studied the Freeswitch doc and can't find any info r
Hi,
One scalability question:
Can someone provide input to the following options?
Option A: Asterisk ( for IVR, Voicemail, Conference) + Openser ( for
SIP registration )
Option B: Freeswitch + Openser
Option C: Freeswitch
So, it is definite that Asterisk is not as scalable as Freeswitch.
Among
Hi,
I have studied the Freeswitch doc and can't find any info related to
setting up the sip users, queue, ans voicemail as a realtime DB, like
Asterisk does. Is this feature available?
If not, would it be a bit too much work to write to the XML all the times?
Also, is there an Asterisk AMI-equi
Hi,
I have been seeing quite a few articles that mention Freeswitch is
much more efficient than Asterisk. What about in terms of
functionality? Can Freeswitch perform the same kind of functionality
provided by Asterisk such as Queue, Conf call, MOH, Voicemail Main,
etc?
Can anyone give some inp
Hi,
This question may have come up a few times already. I am working on
a application to provide IVR, voicemail, and tailored call routing
services. The SIP registration will be handled by Openser, and
Asterisk is only doing the media function. We are talking about over
100 users.
Is Freeswit
Hi,
The application is a IP-PBX with incoming calls through T1 and VOIP
carriers. Asterisk will be used for Voicemail and IVR. I have written some
AGI script to route calls based on user preference which is stored in the
database. Users may connect through softphone or call in through PSTN. Fo
Hi all,
I am currently evaluating either using Freeswitch or Openser for my SIP
server. I am trying to look for information that can tell me the difference
between the two in terms of ease-of-maintain, ease-of-implement, and
scalability. If you have such kind of information, would you please shar
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