current configure will automatically use odbc if it's available, no
need the --enable-core-odbc-support anymore.
better to check if unixodbc-dev package installed of not.
On Jun 16, 2009, at 8:51 PM, bakko wrote:
> Did you compiled freeswitch with this command?
>
> ./configure --enable-core-od
Almost caught you on IRC Mike.
Our server is in a NAT'd network and all agents registered in the same
LAN. I can remotely register by using the public IP and the contact
string is right.
Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE.
User: 6...@192.168.1.16
Contact:
Hi, the difference is the Contact where the "%3B" should be ";" . Is
it configurable or a bug?
13272:
Call-ID:OTg4NmRlNzY5OThmNzgwM2E3ZmRkYzVhNjVmODMyYjA.
User: 8...@192.168.1.15
Contact:"user" >
Agent: Zoiper rev.3065
Status: Registered(UDP)(unknown
how to help without seeing your dialplan?
On Jun 15, 2009, at 6:26 PM, Durk de Beer wrote:
Hello I've minimized de xml files where possible to make a dialplan
that is as short as possible. Now do I've this funny effect to dial
my extensions who are running from 200 to 207. It seams that I'm
On Jun 15, 2009, at 6:21 PM, Edmar Cruz wrote:
>
> Yup tcapi is a great cake php GUI for freeswitch but it is not yet
> fully
> developed...
> Is there any GUI with billing options?
>
>
AFAIK, no fully developed GUI available yet, just curious, why are you
finding a GUI instead of wikipbx or
On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote:
Hi All,
I just finished installing freeSwitch and Skypiax. And I'm able to
use skype api directly via the sk command like the following:
freeswi...@localhost.localdomain>sk console skypiax1
freeswi...@localhost.localdomain>sk CALL userAAA
I
originate sofia/default/1003 &echo()
originate user/1003 &echo()
originate user/1003 &park()
On Jun 10, 2009, at 9:39 PM, Max Bridgewater wrote:
Hi,
Getting to learn Freeswitch. So, please bear with me. Can somebody
tells me how to do dial extension 1003 from the command line.
I tried the
On Jun 10, 2009, at 9:28 PM, Giovanni Maruzzelli wrote:
> On Wed, Jun 10, 2009 at 3:10 PM, dujinfang wrote:
>> btw, I have changed the start scripts a bit to start X and skype
>> separately,
>> glad to share it if someone interested.
>
> I
Glad to see the patch, I have been waiting for a long time :)
btw, I have changed the start scripts a bit to start X and skype
separately, glad to share it if someone interested.
On Jun 10, 2009, at 8:29 PM, Muhammad Shahzad wrote:
I am glad to share the patch to enable dynamic Skypiax inter
2) Is there anyway to have xml_curl send the password for directory
entry requests.
try this:
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nyone knows
how to configure kewlstart with freeswitch/openzap? guess we don't
need tone_detect then.
Dujinfang,
Your telco must support "kewlstart" signaling for this to be
effective. The telco probably calls it something different, like
"disconnect supervision"
yes. Did you ever tried that?
On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote:
I mean when 1001 and 1002 are talking to each other ,
then 1001 want to transfer 1002 to 1003.
2009/6/4 seven
the default config allows 1002 press *1 and 1003 to do blind
transfer, also you may interest the att_xfer,
On Jun 4, 2009, at 9:04 AM, srbrth rsttr wrote:
I don't know how to paste to pastebin, I have registered with
freeswitch.org, but it seems I need a different login for pastebin?
Look closely to the pastebin login prompt box, you will immediately
know how to login.
Meanwhile here's what
On Jun 2, 2009, at 8:01 PM, Juan Backson wrote:
> Hi,
>
> I read the fifo section of the wiki and what is not clear are:
>
> What is the meaning of fifo_orbit_announce?
play to caller before hear the agent.
>
> What is the meaning of fifo_override_announce?
play to agent before hear caller.
>
have you tried this?
http://wiki.freeswitch.org/wiki/Channel_Variables#disable_rtp_auto_adjust
On Jun 3, 2009, at 11:29 PM, Rudolf Denert wrote:
> How can I avoid the problem? It seems that RTP auto adjust generates
> the error. Maybe I can deactivate RTP auto adjust but I suspect that
> fre
I had this problem when I gateway to an asterisk box. Each time I call
to asterisk through that gateway got a 407 and fail. Never figured out
why but guess it's non-proper configuration of Asterisk.
On Jun 3, 2009, at 11:16 PM, bakko wrote:
Ok.
But why i receive a 407 response if no call
I thought it is a problem, made a jira:
http://jira.freeswitch.org/browse/XML-2
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I would route the DID to the host and port 5080 if you are using the
default config, and make an extension in dialplan/public.xml to catch
the DID. Press F8 to see the debug information if not sure what DID
string should be matched.
On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote:
Hello, I h
sorry forgot to mention I'm on FreeSWITCH Version 1.0.trunk (13524M)
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Hi,
I always got 0 messages when using web. Finally I added some debug
information in the code and get this:
2009-06-02 22:20:51 [INFO] mod_voicemail.c:3883
voicemail_api_function() port:[8080]
2009-06-02 22:20:51 [INFO] mod_voicemail.c:3884
voicemail_api_function() uri:[/domains/192.168.1.
Actually Brain mentioned that you can comment out switch_nat_init();
in switch_core.c
On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote:
As I understand it, a new ‘feature’ was added over the weekend to
resolve NAT. If you’re firewall is not allowing ICMP then FS waits
until it times out.
On May 29, 2009, at 11:33 PM, Gerry Hull wrote:
Hi Anthony,
I updated to rev 13496 -- now I have a different problem... I
connect to the event socket interface, ask for all events... then
never receive any events!
From telnet:
"
Content-Type: auth/request
auth ClueCon
Content-Type: com
admutex.o src/esl_config.o
SRC=src/esl.c src/esl_event.c src/esl_threadmutex.c src/
esl_config.c src/esl_oop.cpp
HEADERS=src/include/esl_config.h src/include/esl_event.h src/
include/esl.h src/include/esl_threadmutex.h src/include/esl_oop.h
-SOLINK=-shared -Xlinker -x
+SOLINK=-dynamiclib -Xlinke
Hi,
Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library
On MacOSX 10.5, I can't get ESL for ruby work. make throws error:
sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make rubymod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/Users/
seven/wor
Voila itself is a good name. ;)
VoilaVoIP
On May 22, 2009, at 12:26 PM, Diego Viola wrote:
> Hey guys,
>
> I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
> cool names for my VoIP company, if you know some please tell me :)
>
> Diego
>
> _
time.
/b
On May 18, 2009, at 6:55 PM, dujinfang wrote:
es, FS(13263) send out 482 request merged to my voip client.
I guess, for some reason, FS doesn't respond to the REGISTER, and
when the client start REGISTER again using another call-id, it
merged the request to one. Anyone ever met
09, at 3:37 AM, Brian West wrote:
Is this in regards to FreeSWITCH or something else you're writing?
/b
On May 18, 2009, at 2:34 PM, dujinfang wrote:
On register, sometimes my voip client got SIP/2.0 482 Request merged
sometimes got 200 ok.
482 also means loop detected. my client only has one
On register, sometimes my voip client got SIP/2.0 482 Request merged
sometimes got 200 ok.
482 also means loop detected. my client only has one account logged in
only one place, and no proxy, can I take 482 as 200 OK?
Thanks.
from RFC 3261:
"8.2.2.2 Merged Requests
If the request has n
Even the b leg cdr is enabled it only remember the final state(channel
vars) on the b leg.
At least there are two possible ways to keep tracking all the gateways:
1) don't use '|' separated dial string, use a lua script like this:
session:execute("bridge", dial_string1);
bridg
I'm collecting rtp stat by using xml_cdr, weird that the b leg marked
all packets as cng. Is that a problem?
Here is a real call last more than 15 minutes (codec PCMU). FreeSWITCH
is in the middle, A leg is another FS server and B leg is a PolyLink
Sip Server.
FreeSWITCH version is 13066.
On May 8, 2009, at 9:32 PM, Brian West wrote:
actually I think we can remove the 102 version that was there for
some google talk thing and I don't think we they do that anymore
I'll have to test but ... if we invite to you with mod=30 you have
to do 30 no exceptions as per the iLBC spec.
rev 13240
On Wed, May 6, 2009 at 7:53 AM, dujinfang wrote:
The patch haven't been merged into trunk. It should be as easy as
execute the following command in the FS source code root dir:
patch < /tmp/the_patch_file_name.diff
I will post an example on the wiki when I finished, hope be
On May 6, 2009, at 9:47 PM, Raymond Chandler wrote:
> chenexyee wrote:
>>
>> 1. user A is in conversation with user B, and at this time, a
>> incoming
>> call from user C comes to A, in this case, I want freeswitch to play
>> busytone to C, how to configure?
> you could use the limit app (mod_l
The patch haven't been merged into trunk. It should be as easy as
execute the following command in the FS source code root dir:
patch < /tmp/the_patch_file_name.diff
I will post an example on the wiki when I finished, hope be soon.
On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote:
Hi Seven,
I
Thank you. Even the client is broken, we cannot fix that as we don't
own the code. But we need that client, is that possible to make FS
work around that?
On May 5, 2009, at 6:41 PM, Brian West wrote:
It appears to be a broken client. Your client doesn't ack with the
to tag like zoiper do
On May 4, 2009, at 9:25 PM, Anthony Minessale wrote:
> When you are using on-hook agents, it's presumed to be under low
> call volume, you can just set the agents to get popped
> into the queue in nowait mode so if the caller changed his mind the
> agent will get a hangup. Remember, if there
e job.
On Tue, Apr 28, 2009 at 8:30 AM, dujinfang
wrote:
Ah, right, that works. I had thought the purpose of members is for
sequential hunting. looks I was wrong.
However, add a | sep'ed dial string is hard to do round robin
hunting, as we don't want the first agent always bu
Ah, right, that works. I had thought the purpose of members is for
sequential hunting. looks I was wrong.
However, add a | sep'ed dial string is hard to do round robin hunting,
as we don't want the first agent always busy while others have nothing
to do. It is possible to add/delete members
Again so quick. Thanks. Changed document:
http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources
On Apr 26, 2009, at 10:40 PM, Anthony Minessale wrote:
ok done
set the var to
true / false /
On Sun, Apr 26, 2009 at 7:09 AM, dujinfang
wrote:
Almost perfect. But I think
te what is wrong?
Best regards,
kokoska.rokoska
David Knell napsal(a):
> Add something like
> memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
> after
> char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
> in switch_ivr_play_say.c (line 395)
>
> --Dave
>
>> Thank you
I haven't tested but I guess it's just like other variables and I
documented to here:
http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources
On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:
>
> Thank you very much, Anthony, for such fast solution!
>
> May I ask you - How
Hi,
I made a patch, so skypiax is possible to do a RR hunt besides the
sequential interface ANY.
Usage:
originate skypiax/RR/other_skype_name
sk list
http://jira.freeswitch.org/browse/MODENDP-211
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There is a dp_tools verbose_events can set that flag, you may try to
transfer into a dialplan or use the inline dialplan
try this, not tested.
> originate sofia/gateway/my_gw/u...@domain.com
'verbose_events,playback:foo.wav,echo' inline
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_I
> Here's what it says when I try to connect to the server:
> =
> # ps aux | grep free
> root 3497 0.6 0.7 16912 8212 ?Sl 12:03 0:00
> /usr/local/freeswitch/bin/freeswitch -nc
>
It seems started, I never used a suse, however, can you try this?
#netstat -an | grep 8021
On Apr 4, 2009, at 8:13 AM, Brian West wrote:
First one to give media wins unless you ignore_early_media
/b
Thanks, I tested again. That's exactly what I want except the problem
sometimes the gateway gives (wrong)early_media but fails immediately,
so I have no chance to hear the early m
On Apr 4, 2009, at 7:53 AM, Jason White wrote:
> dujinfang wrote:
>> However, the caller do need to hear the early media to figure out
>> what's going on. If I set ignore_early_media=false, only the first
>> one
>> tried.
>
> Could you use ring_ready?
Hi,
I have outbound gateways returns 403 or 503 constantly. So I tried to
dial out using
sofia/gateways/gw1/|sofia/gateways/gw2/|sofia/gateways/gw3...
for fail over. To make it work, I need to set ignore_early_media=true.
However, the caller do need to hear the early media to figure
On Mar 28, 2009, at 4:05 AM, Anthony Minessale wrote:
if you prefix the sofia dial string with sip: you should be able to
pass anything you want.
sofia/internal/sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060
Is that similar as this?
got it from wiki:
http://wiki.freeswit
ls). Like the
Linux kernel, to compile from source, gcc-3 was recommended for a long
time. Don't know if it's still the case recently.
2009/3/27 dujinfang
On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote:
We do not support ubuntu interpid, it has at least 3 known fatal
On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote:
We do not support ubuntu interpid, it has at least 3 known fatal
issues not experienced by all but nonetheless enough to make us
unwilling to support it.
I use Ubuntu gutsy in production and interipid in test env. It works
well. Can
I don't think you need FREESWITCH_CONFIG. It will find all
configuration files on default place, say /usr/local/freeswitch/conf
If you store config files in other place, the command line should like
this
/usr/local/freeswitch/bin/freeswitch -conf /tmp/conf -db /tmp/db -log /
tmp/log
On Mar
On Mar 25, 2009, at 2:42 AM, Brian West wrote:
How about we all work together and work on the FreeSWITCH.org
infrastructure instead of spreading the resources thinner and
thinner till nobody is doing really much of anything.
I agree with this and would like to work with the wiki.freeswit
On Mar 25, 2009, at 2:03 AM, Michael Collins wrote:
>> While FS official site has plenty of documentation, obviously we
>> don't
>> want to translate word by word. Then where should we start from? A
>> forum?
>
> I would start by finding as many people as possible who are literate
> in both En
Hi all,
In about a year of playing with FreeSWITCH, I really like it. Not only
the software is great but also the community. As people keep asking me
where can find some documentation in Chinese, I told them if they want
to play in deep they need to know English better. However, the fact
i
Hi all,
mod_dingaling in client mode works well for me, but disconnected
yesterday.
2009-03-18 16:57:32 [DEBUG] libdingaling.c:1545 xmpp_connect() io
error 2 7
I use dl_login profile=gmail.com, and it re-login successfully. Is
their a way to auto re-login after fail?
Thanks.
Maybe it can help by following this thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012083.html
On Mar 17, 2009, at 11:23 PM, Christian Benke wrote:
> Hi!
>
> Is this not possible with registration at a gateway or is there a
> other
> reason why i didn't get any respo
at default config, in conf/sip_profiles/external.xml
where $${external_sip_port} is a variable you can find in conf/
vars.xml, normally it's 5080, make sure sipbroker route calls to that
port, and then you can make a dialplan in
conf/dialplan/public.xml
turn on verbose log
I only use skypiax do outbound calls by using the ANY interface, and
it works pretty well. It will be really cool if multi channels can
call in to a single account. However, AFAIK, the skype client on my
computer, if the second call come in the first channel will change to
"hold". It does c
We are using ubuntu 8.04 in Xen(also hosted by ubuntu 8.04, Ubuntu
8.10 is not xen friendly) as our testing server. It works well,
however we only use that to test our business logic, not press test at
all.
On Mar 6, 2009, at 9:01 PM, EdPimentl wrote:
> Anyone using uBuntu 8.10 and XEN?
> W
How about this?
bridge
({origination_caller_id_name
="your_name",origination_caller_id_number=""}sofia/b-leg)
On Mar 6, 2009, at 3:51 PM, rod wrote:
> using these functions like this did nothing on the SIP INVITE
> packet :'(
>
> seven wrote:
>> try
>> bridge
>> ({effective_caller_id_name
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