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
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.
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
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
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
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
-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 related to
setting up the sip users, queue, ans
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
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,
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
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
it should actually be:
anti-action application=bridge data=sofia/default/$
{dialed_ext}%$${domain}/
Check
http://wiki.freeswitch.org/wiki/Sofia#Call_a_locally_registered_endpoint
Cheers,
UV
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]
] On Behalf Of Pete Kay
Sent
Hi,
I comment the line at default.xml
!--anti-action application=set
data=sip_exclude_contact=${network_addr}/ --
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]
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
Hi,
I place my xml response generator directory.php under
/var/www/fs/directory.php
In my xml_curl_conf.xml, I have
binding name=directory fetcher
param name=gateway-url value=http://localhost/fs/;
bindings=directory.php/
/binding
2008-04-18 22:23:17 [ERR] mod_xml_curl.c:251
.
Thanks,
Pete
On Fri, Apr 18, 2008 at 2:28 PM, Pete Kay [EMAIL PROTECTED] wrote:
Hi,
I place my xml response generator directory.php under
/var/www/fs/directory.php
In my xml_curl_conf.xml, I have
binding name=directory fetcher
param name=gateway-url value=http://localhost/fs
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,
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?
? 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 Apr 18, 2008, at 6:25 AM, Pete Kay wrote:
2008
Hi,
I two a few questions about hangup and sounds:
1.
In my dialplan, I have
extension name=catchall
condition field=destination_number expression=^(.*)$
action application=set data=default_language=zh/
action application=phrase data=call-back-later/
action
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
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
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:
action application=voicemail data=auth default 1005/
Instead, it goes to the mailbox asking user to record message:
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,
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
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
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 a
27 matches
Mail list logo
