Hello,
I can't get call transfer to work with a SPA2102 adapter.
I don't think it has something to do with FS, but I'm hoping someone here
could help me.
I do not get a new line in the phone (by pressing the R button), all DTMF
tones are sent as audio to the other connected phone.
Anyone got it
I have the same problem with a HandyTone 502 adapter.
Anyone got any hints to get the flash button to work?
On Wed, Dec 9, 2009 at 11:25 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
Hello,
I can't get call transfer to work with a SPA2102 adapter.
I don't think it has something to do
I got a freeswitch that is behind nat and got three profiles.
External (all calls are going through a proxy):
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip
...@freeswitch.org wrote:
Are you doing this all on a linux box thats acting as your router too? If
not you don't need two profiles... you also don't need to set the
local-network-acl on ANY profile that isn't do anything with nat.
/b
On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote:
I got a freeswitch
:
In this case you should not need 2 profiles either.
On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote:
It's a windowsserver which is behind a router.
Which profile should local-network-acl be specified on?
When I bridge calls to the outside world, should I use
sofia/internal/phoneNumber
Hello
I got the following setup: Phones - FreeSwitch - NAT - Internet -
Gateway
And I'm struggling to get NAT working properly. I'm running freeswitch with
the -nonat option and have tried different ext-rtp-ip/ext-sip-ip
combinations in external/internal profiles.
The From header seems to be
instead of effective_caller_id?
On Mon, Nov 23, 2009 at 6:08 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
Hello
I got the following setup: Phones - FreeSwitch - NAT - Internet -
Gateway
And I'm struggling to get NAT working properly. I'm running freeswitch with
the -nonat option and have
:
outbound_caller_id is a made up variable that is used in the defaults that
are used in the examples only.
/b
On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote:
Ok. Found the problem. I had started using sofia/outbound/
xxx...@sipgw2..se as bridge destination to try to get
Hello,
What should my acl.conf.xml look like if I want to allow ALL calls on the
external profile and use only digest authentication on all other profiles?
Thanks,
Jonas
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switch_ivr_play_say.c, line 486.
file = switch_core_session_sprintf(session, %s%s%s%s,
switch_str_nil(tfile), tfile ? ] : , prefix, SWITCH_PATH_SEPARATOR,
file);
There should be five %s, not four.
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Same bug in switch_ivr_async.c, method switch_ivr_record_session.
On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
switch_ivr_play_say.c, line 486.
file = switch_core_session_sprintf(session, %s%s%s%s,
switch_str_nil(tfile), tfile ? ] : , prefix
Hello,
I have to static IP:s on my server. FS has been bound to one of them.
Yesterday evening I got these log messages:
2009-05-05 19:17:37 [INFO] mod_sofia.c:2938 general_event_handler() IP
change detected [85.89.XX.XX9]-[85.89.XX.XX8] []-[]
2009-05-05 19:17:37 [NOTICE] sofia_glue.c:3303
you happen to bind the IP while FS was running?
/b
On May 6, 2009, at 1:36 AM, Jonas Gauffin wrote:
Hello,
I have to static IP:s on my server. FS has been bound to one of them.
Yesterday evening I got these log messages:
2009-05-05 19:17:37 [INFO] mod_sofia.c:2938 general_event_handler
I'm using session.recordFile in a javascript.
How do I check the length of the recorded file?
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Yes, to be able to determine if it's too short or not.
On Tue, Apr 21, 2009 at 7:01 PM, Michael Collins m...@freeswitch.org wrote:
On Tue, Apr 21, 2009 at 4:23 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
I'm using session.recordFile in a javascript.
How do I check the length
How?
On Tue, Apr 21, 2009 at 8:02 PM, Stephen Crosby stevecr...@gmail.comwrote:
You can make a really close estimation based on the size of the recorded
file. That's what I've been doing.
--Stephen
On Tue, Apr 21, 2009 at 10:58 AM, Jonas Gauffin
jonas.gauf...@gmail.comwrote:
Yes
Thanks! :)
On Tue, Apr 21, 2009 at 9:34 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
update to trunk and you should have
record_ms and record_samples chanvars
also playback_ms and playback_samples for playing
On Tue, Apr 21, 2009 at 12:58 PM, Jonas Gauffin
jonas.gauf
=200 rtd=2
/recv
ResponseTimeRepartition value=10, 20, 30, 40, 50, 100, 250, 500/
/scenario
2009/3/26 Jonas Gauffin jonas.gauf...@gmail.com
Hello
I want to achive this: Sipp1 - FS - Sipp2
Sipp1 emulates a inbound calls (easy to achive)
Sipp2 should emulate a registered user (i.e
Hello
I want to achive this: Sipp1 - FS - Sipp2
Sipp1 emulates a inbound calls (easy to achive)
Sipp2 should emulate a registered user (i.e. register with FS and then just
wait for calls and hangup when sipp1 hangsup)
How do I configure sipp as Sipp2?
Thanks,
Jonas
Hello
I'm trying to make outbound calls through my gateway provider.
My calls got rejected and I asked them why.
Apparently I need to use 5060 as source port, since they validate both my IP
and the port that the messages come from.
Is this possible with freeswitch? If so, what config settings
port on every packet will be 5060.
If you *are* behind nat the nat mapping will pick a random port unless you
have a firewall that allows you to set specific mappings.
On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
Hello
I'm trying to make outbound calls through my
created that way and not the session thread.
On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
Hello
I got problems with hanging spidermonkey sessions and need some advice on
how to debug them.
I've made a javascript queue application that uses
it.
Will new sessions always wait on old ones to be garbage collected properly?
For instance, what happens if a script have a lenghty post process after
caller have hang up?
On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
I've got a loop, but the first thing checked
Hello
I got problems with hanging spidermonkey sessions and need some advice on
how to debug them.
I've made a javascript queue application that uses mod_spidermonkey_socket.
It works fine for a while,
but after some calls I noticed that calls didnt get transferred to agents.
The reason was that
I'm trying to build latest trunk on win (vs2008) and get the following
errors:
12mod_spidermonkey.obj : error LNK2001: unresolved external symbol
_switch_dso_open
12mod_spidermonkey.obj : error LNK2001: unresolved external symbol
_switch_dso_data_sym
any thoughts?
thanks mate.
On Wed, Jan 7, 2009 at 3:26 PM, Michael Jerris m...@jerris.com wrote:
New Revision: 11085
Log: fix windows build breakage from svn rev 11084
On Jan 7, 2009, at 6:18 AM, Jonas Gauffin wrote:
I'm trying to build latest trunk on win (vs2008) and get the
following errors
PCMU in the conf.
On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin
jonas.gauf...@gmail.comwrote:
Hello
I got problems with choppy voice. I just happens1 time of 10 or something
like that.
Incorrect call: http://pastebin.freeswitch.org/6479
working call: http://pastebin.freeswitch.org/6481
/section
/document
That's it. the sofia configs are pretty much default.
On Mon, Dec 15, 2008 at 5:53 PM, Raymond Chandler
intralan...@freeswitch.org wrote:
posting relevant pieces of your dialplan and sofia configs would probably
help a bit.
-Ray
Jonas Gauffin wrote
I get a bind error for the RTP, can someone be kind and explain why?
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2388 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000]
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1596 sofia_glue_tech_set_codec()
Set Codec
15, 2008, at 5:03 AM, Jason White wrote:
On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
I'm trying to bridge using a non-registered gateway. And I
get MANDATORY_IE_MISSING back. Why is that?
Does the gateway allow unauthenticated clients to make calls? If you
obtain
I'm trying to bridge using a non-registered gateway. And I
get MANDATORY_IE_MISSING back. Why is that?
2008-12-15 11:06:49 [DEBUG] switch_core_state_machine.c:152
switch_core_standard_on_execute()
sofia/internal/u1000...@192.168.1.112:5070Execute
bridge(sofia/default/
]sofia/foo/[EMAIL PROTECTED]
On Wed, Nov 26, 2008 at 9:20 AM, Jonas Gauffin [EMAIL PROTECTED]wrote:
Hello
I send an API command through the event socket that looks like this (the
first two variables are used by our server):
api originate
{gate_user_id=44,gate_site_id=1,sip_invite_params
Hello
Do anyone know where I can find a xml/csv file with international/national
dialing codes?
Thanks,
Jonas
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Hi Mike,
This is a problem for me too. I did a fresh checkout yesterday.
Regards,
Jonas
On Tue, Oct 28, 2008 at 10:06 AM, Tamas Cseke [EMAIL PROTECTED]wrote:
Hello,
I still have problem after a fresh checkout.
I tried with MSVC++ 2008 express and I got the same errors too.
Tamas
Ok, the 2008 version worked.
But in the 2008 version, the setup project is not included (in the solution
file).
I added it and compiled it, and got these errros:
-- Starting pre-build validation for project 'FreeSwitchSetup' --
ERROR: Unable to find source file
Hello
What kind of system requirements do FS have to handle about 500 simultaneous
calls?
The specsheet in the wiki have Minimum/Recommended System Requirements, but
they just talk about memory and disk space (and doesnt mention how many
calls the recomendations are for).
Thanks,
Jonas
doh. It was too close :)
On Wed, Oct 29, 2008 at 10:40 AM, ram [EMAIL PROTECTED] wrote:
On Wed, Oct 29, 2008 at 1:03 PM, Jonas Gauffin [EMAIL PROTECTED]wrote:
Hello
What kind of system requirements do FS have to handle about 500
simultaneous calls?
The specsheet in the wiki have
Is it possible to disable this check?
On Wed, Oct 8, 2008 at 3:24 PM, Jonas Gauffin [EMAIL PROTECTED]wrote:
It have happened once on that server, and once on another one which uses
DHCP (but the IP have not changed since I installed the server).
On Wed, Oct 8, 2008 at 3:15 PM, Anthony
PROTECTED] wrote:
Their might be cases this is suboptimal but have you seen such a case
yet? If we don't bounce then the profiles won't work if the ip is
yanks out from under it.
/b
On Oct 17, 2008, at 12:00 AM, Jonas Gauffin wrote:
Is it possible to disable this check
of the feature would be excellent for me.
On Fri, Oct 17, 2008 at 10:31 AM, Brian West [EMAIL PROTECTED] wrote:
Can you help narrow down what is going on to trigger it when it really
doesn't take place?
/b
On Oct 17, 2008, at 1:27 AM, Jonas Gauffin wrote:
Well. My case is that the IPs have
On the server with two static ips then?
On Fri, Oct 17, 2008 at 10:49 AM, Brian West [EMAIL PROTECTED] wrote:
The DHCP interface must have changed...
/b
On Oct 17, 2008, at 1:39 AM, Jonas Gauffin wrote:
I got two dedicated FS servers and it have happened once on each
server.
One
Hello
How do I specify callerid in the dialplan?
I tried with:
action name=set data=caller_id_name=Test Telefon /
action name=set data=caller_id_num=XX /
action name=bridge data=sofia/default/[EMAIL PROTECTED] /
The gateway still receives the callerid that I've specified in the user
thanks
On Fri, Oct 17, 2008 at 3:23 PM, Tamas Cseke [EMAIL PROTECTED]wrote:
action name=set data=effective_caller_id_name=X/
action name=set data=effective_caller_id_number=X/
I guess
Jonas Gauffin írta:
Hello
How do I specify callerid in the dialplan?
I tried
your DSN name is postgres and not PostgreSQL
On Fri, Oct 10, 2008 at 12:34 PM, Gayatri Kulkarni
[EMAIL PROTECTED]wrote:
Guys please help me on this
I am still getting that error
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0
Hello
Today Freeswitch restarted the profiles, and I don't know why.
2008-10-08 09:44:49 [NOTICE] sofia_glue.c:2634
sofia_glue_restart_all_profiles() Reload XML [Success]
2008-10-08 09:44:49 [DEBUG] sofia.c:634 sofia_profile_thread_run() Write
lock outbound
2008-10-08 09:44:49 [NOTICE]
Hello
I got the following setup:
Phone1 and FS behind NAT1.
Phone2 behind NAT2
Phone1 - FS - NAT1 - INTERNET - NAT2 - Phone2
Both phone1 and phone2 registers to FS. Phone1 on the internal profile
and Phone2 on a profile that is identical to internal, except that
external sip/rtp ips are set.
on a public IP, or behind nat?
Ivan
Den 16. sep.. 2008 kl. 13:44 skrev Jonas Gauffin:
Hello
I haven't touched my sofia profiles in a long time.
I now need to have two profiles for phones registering against
FreeSWITCH:
* One for phones that are on the same LAN as FS
* One
Hello
Is it possible to do something like this (pseudo code):
set call_timeout=20
bridge somehere
if timeout bridge somewhereElse
else javascript unexpectedFailure.js
that is invoke one action if the bridge fails due to call timeout, and to do
something else for every other reason.
//Jonas
Hello
I'm running FS as a NT service. Is it possible to access the FS console
through telnet or something like that?
//Jonas
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Hello
What does mod_esf and mod_fsv do?
mod_esf just says Extra SIP Functionality in the wiki. mod_fsv is not
documented at all.
Regards,
Jonas
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I had similar problems:
http://jira.freeswitch.org/browse/MODEVENT-25
Can you determine if your problem is the same?
On Mon, Sep 8, 2008 at 4:38 PM, Erol Akarsu [EMAIL PROTECTED] wrote:
I am having a issue on CHANNEL_DESTROY event.
I have sent 20 originate calls and receiving only 19
err. it should say ... and I have NOT tested the merged fix yet.
On Mon, Sep 8, 2008 at 5:00 PM, Jonas Gauffin [EMAIL PROTECTED]wrote:
The patch was merged with another fix, and I have tested the merged fix
yet.
I'm still running with my patch (which is attached to MODEVENT-25
[tablename] |
| quit |
| |
+---+
SQL
Jonas Gauffin wrote:
First you have to add
use(ODBC);
in your javascript.
On Thu, Aug 28, 2008 at 9:06 AM, mashudi [EMAIL
First you have to add
use(ODBC);
in your javascript.
On Thu, Aug 28, 2008 at 9:06 AM, mashudi [EMAIL PROTECTED] wrote:
Hi folks,
i try javascripts example for conference application and get this error,
2008-08-28 20:59:55 [ERR] mod_spidermonkey.c:3303 js_api_use() Error
loading ODBC
Which codec are you using?
On Thu, Aug 21, 2008 at 8:57 AM, Ilan Perez [EMAIL PROTECTED] wrote:
How can I improve the sound quality that FS records?
Ilan Perez
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Show us logs of a call attempt.
On Thu, Aug 21, 2008 at 8:42 AM, Adeel Ansari [EMAIL PROTECTED] wrote:
5080 and 5060 both are open. Moreover, other PBX is working fine on 5060.
Further, I have tried by setting the parameters sip-ip and rtp-ip to auto;
ext-sip-ip and ext-sip-ip to my public
EXCHANGE_MEDIA
2008-08-21 15:15:58 [DEBUG] mod_sofia.c:365 sofia_on_exchange_media() SOFIA
LOOPBACK
==
Thanks.
On Thu, Aug 21, 2008 at 3:07 PM, Jonas Gauffin [EMAIL PROTECTED]
wrote:
Show us logs of a call attempt.
On Thu, Aug 21, 2008 at 8:42 AM, Adeel Ansari [EMAIL PROTECTED]
wrote
,
nothing changed same local ip. I have tried to give ext-rtp-ip my public ip,
but no work. Any suggestions?
Thanks for your efforts.
On Thu, Aug 21, 2008 at 3:26 PM, Jonas Gauffin [EMAIL PROTECTED]
wrote:
None of the phones are on the same lan as freeswitch, right?
If so, one
switch_ivr_play_say.c in the core (function: switch_ivr_record_file)
On Thu, Aug 21, 2008 at 9:50 AM, Ilan Perez [EMAIL PROTECTED] wrote:
Where can I find the code in FS tha deals with the actual recording…
Not the xml code to record a session but the pro-compiled record function?
Ilan
Hello
Is it possible to use sip_auto_answer when sending an originate
command through the event socket?
Regards,
Jonas
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in the default dialplan it will work.
/b
On Aug 21, 2008, at 10:57 AM, Jonas Gauffin wrote:
Hello
Is it possible to use sip_auto_answer when sending an originate
command through the event socket?
Regards,
Jonas
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Are max_forwards being used?
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Here is how I do it:
var s = new Session();
s.setCallerData(caller_id_name, Inspelning);
s.setCallerData(caller_id_number, inget);
s.originate(session, destination);
session = s;
where destination is something like sofia/yourdomain.com/destination
On Tue, Aug 19, 2008 at 12:56 PM, James Green
does it work without SSL?
On Tue, Aug 19, 2008 at 2:24 PM, kokoska rokoska
[EMAIL PROTECTED] wrote:
Hi all,
I try to use mod_xml_cdr for posting CDR records to my web server but
without luck. The situation is as follows:
1. I utilize mod_xml_curl to serve configuration, directory and
I had the same problem a while ago. I don't know if it's fixed, but I
do this (and it works)
session.setVariable(ringback,
%(1000,4000,425));
bleg = new Session();
bleg.setCallerData(caller_id_name,
Does this apply to:
http://wiki.freeswitch.org/wiki/Session_waitForAnswer
http://wiki.freeswitch.org/wiki/Session_originate
I can update the wiki if so.
On Tue, Aug 19, 2008 at 3:18 PM, Brian West [EMAIL PROTECTED] wrote:
Using setCallerData isn't the proper way to set the callerid.
Is it possible to get FS to just report open/closed (avail/not avail)
instead of all specific call states?
Maybe a toggle for that or something?
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I'm working on one, since I've rebuilt it a bit.
On Mon, Aug 4, 2008 at 3:48 PM, Gerry Hull [EMAIL PROTECTED] wrote:
Does anyone have one?
Gerry
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since the port is no longer active and
the GS just keeps trying to register on a bad port. The only fix is to
reset the GS every time FS is restarted.
Jonas Gauffin wrote:
Hello
I have problems with TCP and GXP2000 phones.
They work fine using UDP, but no calls arrive if I switch to TCP.
I've
Hello
I have problems with TCP and GXP2000 phones.
They work fine using UDP, but no calls arrive if I switch to TCP.
I've upgraded to the latest GS firmware today to see it that helped,
but it didn't.
Have anyone else had problems with TCP?
Regards,
Jonas
I've started to do a mod_ivr_socket a long time ago, which I never completed.
It should be considered as an alternative to the scripting languages
and gives you an easier interface than the event socket.
I created it since I wanted to move the ivr applications from my
javascripts into my own
At home. I'll try to add it this weekend.
On Fri, Aug 1, 2008 at 4:12 PM, Erol Akarsu [EMAIL PROTECTED] wrote:
Jonas,
This is what I want.
Do you have any documentation on mod_ivr_socket and how its API looks like?
Thanks
Erol Akarsu
- Original Message
From: Jonas Gauffin
Hello
What should I add in the diaplan if I want to reject a call with busy
as status code?
Should I add the hangup application with SWITCH_CAUSE_BUSY or the
respond application with 486?
//Jonas
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, Jonas Gauffin wrote:
Hello
I got some problems with the event socket, I do not receive all
events.
I've confirmed this by adding logs to mod_event_socket (line 548):
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, Sending
event: %s\n, switch_event_get_header(pevent, Event-Name));
len
http://www.opensourcesip.org:8080/clearspacex/community/opensbc
there you got a open source sbc.
On Wed, May 14, 2008 at 5:22 AM, mashudi [EMAIL PROTECTED] wrote:
Dear all,
is it possible to configure FS as SBC ? so it can connect two different
network ?
Hello
Is it possible to get support for response code 484 in mod_xml_curl dialplan?
That is to return status code address incomplete instead of not
found in the xml?
//Jonas
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Doesn't that just allow media to be sent before a 200 ok? In that
case, it doesn't solve the waitForAnswer problem?
On Fri, May 9, 2008 at 7:05 AM, Leonardo Alves [EMAIL PROTECTED] wrote:
Try using the parameter early_media=true
From: Marc Orenberg
Sent: Thursday, May 08, 2008 9:39 PM
To:
Hello
Someone was interested in a EventSocket implementation in .Net.
I've added by version to the tree now. It can be found in
freeswitch\scripts\contrib\verifier.
I've also added it to codeplex, http://www.codeplex.com/eventsocket,
will also add a binary there in a few days.
I've also started
If think that I had done most of it, but i havent tested it though.
http://jira.freeswitch.org/browse/MODEVENT-23
On Fri, Apr 25, 2008 at 2:39 PM, Yossi Neiman
[EMAIL PROTECTED] wrote:
Jonas Gauffin wrote:
I've done a mod_odbc module that writes to a flatfile if odbc fails
(and adds them
collect the sip trace of
the redirect?
/b
On Apr 21, 2008, at 9:31 AM, Jonas Gauffin wrote:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=dialplan description=ModCurlDialplan
context name=default
extension name
at the default config at the unroll loops extension... that is
what processes the loop.
/b
On Apr 21, 2008, at 6:04 AM, Jonas Gauffin wrote:
Hello
I'm trying to get redirect to work in the dialplan.
Here's the actual attempt:
13:00:18 [DEBUG] mod_sofia.c:119 sofia_on_ring
just trying to send a 302.
On Mon, Apr 21, 2008 at 3:55 PM, Brian West [EMAIL PROTECTED] wrote:
Is this a redirect done by the far end device? or you're just trying
to send a 302 from FreeSWITCH?
/b
On Apr 21, 2008, at 8:47 AM, Jonas Gauffin wrote:
huh?
I'm using a custom
you? That is normal the codec
for the call looks to be PCMU but to play the file it needs L16 so the
core can translate it to the caller. Plus you seem to have not posted
the full trace so I can see.
console loglevel 8
and paste that.
/b
On Apr 16, 2008, at 1:35 AM, Jonas Gauffin
Hello again =)
I've made a call queue in javascript where the js takes orders from a
server through a socket.
The server monitors all extensions (through eventsocket) in the queue
and tells the javascript to originate the call to a extension as soon
as it becomes idle.
Everything works fine
(execute_extenson, 1000);
Which unlike transfer runs the extension inline and returns back to the same
place.
On Wed, Apr 16, 2008 at 8:29 AM, Jonas Gauffin [EMAIL PROTECTED]
wrote:
Hello again =)
I've made a call queue in javascript where the js takes orders from a
server
http://jira.freeswitch.org/browse/MODLANG-58
On Wed, Apr 16, 2008 at 4:23 PM, Jonas Gauffin [EMAIL PROTECTED] wrote:
No, I'm creating a new session:
bleg = new Session();
bleg.setCallerData(caller_id_name,
session.caller_id_name
http://jira.freeswitch.org/browse/MODLANG-56
On Wed, Apr 16, 2008 at 4:15 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
Can you please file it on jira and attach the script to reproduce it and any
relevant logs?
On Wed, Apr 16, 2008 at 8:17 AM, Jonas Gauffin [EMAIL PROTECTED]
wrote
Hello
I get a debug assert failure when using swedish characters in caller id name.
File: isctype.c
Line: 56
Expression: (unsigned)(c + 1) = 256
The character i tried with is ö (o with two dots over if mailinglist
cant handle it, html encoding: ouml;).
Provided directory from my (local
detailed back trace of the issue so we can follow it up
to our code and see if we can prevent it.
On Tue, Apr 15, 2008 at 10:22 AM, Jonas Gauffin [EMAIL PROTECTED]
wrote:
Hello
I get a debug assert failure when using swedish characters in caller id
name.
File: isctype.c
Line: 56
Jerris [EMAIL PROTECTED] wrote:
I think you will need to try to tweak the build for unicode support to
fix this.
Mike
On Apr 15, 2008, at 11:48 AM, Jonas Gauffin wrote:
assert is made on while (*s *s != '' (*s != '%' || t != '%')
!isspace((int) (*s)))
*s contains ösvan
Hello
I'm launching a javascript that originates a call to a user and then
records a file.
The problem is that the soundfiles (record after the beep and
message saved) sounds strange (choppy/vibrating, yeah I know, sucky
explanation).
The same phrases sounds fine when played for an incoming
Hello
som phones have different fields for auth username and username.
Freeswitch sends the auth-username when requesting the directory, but
then registers the user using the regular username.
For me this is a problem, since I generate dialplans dynamically in my
app using the info sent in the
note that reloading a profile will abort all active calls in that profile.
On Mon, Apr 7, 2008 at 7:52 PM, David Knell [EMAIL PROTECTED] wrote:
Hi Kukoska,
I did this using mod_xml_curl, which worked fine: you also need to poke
appropriate commands in to FreeSWITCH to have it reload the
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