Is there any example on how to use mod_fifo?
I am trying to implement a call centre queue as follows (much like
Asterisk queues) :
Inbound call-> press 0 for operator -> mod_fifo -> 3 agents of whom any
one can get the call (doing round robin or whatever)
I checked out:
http://wiki.freeswitch.
The comment typo was already fixed, sorry, not sure how that even
ended up in there.
Mike
On Jul 1, 2008, at 10:01 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Can you open an Jira on this so we don't loose track of it..
> http://jira.freeswitch.org
>
> /b
>
> On Jul 1, 2008, at 8:29 PM, John
Can you open an Jira on this so we don't loose track of it..
http://jira.freeswitch.org
/b
On Jul 1, 2008, at 8:29 PM, John Wehle wrote:
>> Can you try upgrading to the latest svn build of FS.
>> There are several fixes to openzap in there that I know will fix
>> your issue.
>
> Yes, that wor
> Can you try upgrading to the latest svn build of FS.
> There are several fixes to openzap in there that I know will fix your issue.
Note I do get the following messages:
[WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:2
[WARNING] zap_zt.c:642 zt_next_event() Unhandled ev
> Can you try upgrading to the latest svn build of FS.
> There are several fixes to openzap in there that I know will fix your issue.
Yes, that works much better. A couple minor changes required so the
openzap code would compile on FreeBSD 6.2:
a) The header for select needed to be included.
Please take the time and make sure the wiki is updated so others can
learn also. I know some of this info is there already but the
usability factor is wrong.
/b
On Jul 1, 2008, at 4:22 PM, Henk Oegema wrote:
> On Tuesday 01 July 2008 23:10:18 Brian West wrote:
>> You'll need to restart the
On Tuesday 01 July 2008 23:10:18 Brian West wrote:
> You'll need to restart the sip profile.
>
> sofia profile external restart reloadxml
>
> The reloadxml arg is option it'll just make sure it reloads the XML
> before it restarts the profile.
Thanks Brian. That works !
I 've learned enough for t
You'll need to restart the sip profile.
sofia profile external restart reloadxml
The reloadxml arg is option it'll just make sure it reloads the XML
before it restarts the profile.
/b
On Jul 1, 2008, at 4:05 PM, Henk Oegema wrote:
> I've added a new ITSP
> to /usr/local/freeswitch/conf/sip_
I've added a new ITSP
to /usr/local/freeswitch/conf/sip_profiles/external/voipraider.xml
(voipbuster works now)
What do I need to reload now to make voipraider visible in 'sofia status' ?
I know that if I stop and start FS it will be there, but I want to simulate a
business environment, were I
No restarting the sip profile.
On Jul 1, 2008, at 3:44 PM, Peder @ NetworkOblivion wrote:
>
> Do you mean doing a reloadacl will cause that? Or a reloadxml in
> general will cause all calls on the profile to drop?
Brian West
sip:[EMAIL PROTECTED]
___
"Please note that doing this will make all the calls on the profile drop."
Do you mean doing a reloadacl will cause that? Or a reloadxml in
general will cause all calls on the profile to drop?
Brian West wrote:
> You can reload the XML via "reloadxml" but many things will not reload
> or ch
Start with this command: reloadxml
That's good for lots of things, especially dialplan changes.
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of Henk Oegema
> Sent: Tuesday, July 01, 2008 1:31 PM
> To: freeswitch-users@lists.freeswit
reloadxml
Henk Oegema wrote:
> When I make a change in a XML file, must I reload (how?) or stop and start
> FS ? Or don't I need to do anything ? (wait for a while ?)
>
> Rgds
> Henk
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists
You can reload the XML via "reloadxml" but many things will not reload
or change automatically this includes all the sip profiles. You can
shut FS down and start it back up or restart the sip profile:
sofia profile restart
ACL's can be reloaded like this
"reloadacl reloadxml" or a reloadxm
When I make a change in a XML file, must I reload (how?) or stop and start
FS ? Or don't I need to do anything ? (wait for a while ?)
Rgds
Henk
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailm
On Tuesday 01 July 2008 21:25:12 David Knell wrote:
> I'd guess that the spare } in here is probaly responsible:
You're right David. My mistake.
>
> 2008-07-01 21:05:04 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
> conditions destination_number(0031344641353) =~ /^(0031\d**}*)$/
>
On Tuesday 01 July 2008 21:20:46 Gonzalo Servat wrote:
> On Tue, Jul 1, 2008 at 4:07 PM, Henk Oegema <[EMAIL PROTECTED]>
>
> Regex problem. You've got:
>
>
>
> You probably want:
>
>
>
> I changed the '*' to a '+' as you probably want at least one digit after
> the 0031 to match, but that's up to
I'd guess that the spare } in here is probaly responsible:
2008-07-01 21:05:04 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions destination_number(0031344641353) =~ /^(0031\d**}*)$/
-Dave
Your regular expression is wrong for your bridge action. Or you put
it in the wrong place.
On Tue, Jul 1, 2008 at 4:07 PM, Henk Oegema <[EMAIL PROTECTED]>
wrote:
> [..snip..]
> 2008-07-01 21:05:04 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
> Processing
> 1000->[EMAIL PROTECTED]
> 2008-07-01 21:05:04 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
> conditions destination_number(003
Your regular expression is wrong for your bridge action. Or you put
it in the wrong place. So it gives up and send it to enum and has no
way to route the call.
/b
On Jul 1, 2008, at 2:07 PM, Henk Oegema wrote:
switch_core_standard_on_execute() sofia/internal/[EMAIL PROTECTED]
Execute
t
On Tuesday 01 July 2008 20:45:57 Brian West wrote:
> Now make the call after this and see what it says.
>
[EMAIL PROTECTED]> 2008-07-01 21:05:04 [NOTICE] switch_channel.c:533
switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED]
[9cc92562-47a0-11dd-88b8-7557dafe7bf1]
2008-07-01 2
Now make the call after this and see what it says.
/b
On Jul 1, 2008, at 1:37 PM, Henk Oegema wrote:
>
> When I press F8, I see:
>
> [EMAIL PROTECTED]>
> API CALL [console(loglevel 7)] output:
> +OK console log level set to DEBUG
>
> [EMAIL PROTECTED]>
Brian West
sip:[EMAIL PROTECTED]
_
On Tue, Jul 1, 2008 at 3:37 PM, Henk Oegema <[EMAIL PROTECTED]>
wrote:
> [..snip..]
> When I press F8, I see:
>
> [EMAIL PROTECTED]>
> API CALL [console(loglevel 7)] output:
> +OK console log level set to DEBUG
>
> [EMAIL PROTECTED]>
>
I'm pretty sure Brian meant "Press F8, try to dial again and
On Tuesday 01 July 2008 20:17:43 Brian West wrote:
> Where did you put your gateway xml for voipbuster?
in /usr/local/freeswitch/conf/sip_profiles/external/
File: voipbuster.xml
-
> Also press F8 and
> see what it says or "console loglevel 8"
Where did you put your gateway xml for voipbuster? Also press F8 and
see what it says or "console loglevel 8"
/b
On Jul 1, 2008, at 1:00 PM, Henk Oegema wrote:
On Tuesday 01 July 2008 19:42:13 Brian West wrote:
> Which profile is your outbound registration going out on?
How do I check th
On Tuesday 01 July 2008 19:42:13 Brian West wrote:
> Which profile is your outbound registration going out on?
How do I check that ?
(sorry Brian, but I'm an absolute beginner)
>
> /b
>
> On Jul 1, 2008, at 12:31 PM, Henk Oegema wrote:
> > 2008-07-01 19:25:54 [INFO] mod_dialplan_xml.c:222 dialpl
Which profile is your outbound registration going out on?
/b
On Jul 1, 2008, at 12:31 PM, Henk Oegema wrote:
2008-07-01 19:25:54 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Processing 1000->[EMAIL PROTECTED]
Brian West
sip:[EMAIL PROTECTED]
_
On Tuesday 01 July 2008 17:52:59 Anthony Minessale wrote:
> try taking out the @voipbuster.com
>
> all you need is
> sofia/gateway/voipbuster/$1
>
I changed it to:
but I'm still not successfull when dialing from internal extension 1000 to
003134464135x
try taking out the @voipbuster.com
all you need is
sofia/gateway/voipbuster/$1
On Tue, Jul 1, 2008 at 10:49 AM, Henk Oegema <[EMAIL PROTECTED]>
wrote:
> I have created a gateway to Voipbuster in
> conf/sip_profiles/external/voipbuster.xml:
>
>
>
>
>
>
>
>
>
>
>
> I suppose this i
I have created a gateway to Voipbuster in
conf/sip_profiles/external/voipbuster.xml:
I suppose this is correct since I can register with Voipbuster:
[EMAIL PROTECTED]> sofia status
API CALL [sofia(status)] output:
Name Type
Can you try upgrading to the latest svn build of FS.
There are several fixes to openzap in there that I know will fix your issue.
BTW
Let me know how it's going, we have not actually seen anyone try openzap on
BSD before.
On Mon, Jun 30, 2008 at 7:29 PM, John Wehle <[EMAIL PROTECTED]> wrote:
>
Diego,
> This is sad, you claim that FreeSWITCH is more advanced in terms of
> features and innovation than Asterisk, but all I see with your
> attitude is that it's more limited and closed-minded.
how come they are close-minded and limited, when open source(e.g
freeswitch) gives you freedom to
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