I created a profile name external5090 on
/usr/local/freeswitch/conf/sip_profiles/external5090.xml... Change
ext-sip-ip and ext-rtp-ip for a server 192.168.0.104 with sip-port: 5090...
My local Ip is 192.168.0.105... I see it with I type it on the API
freeswitch and type sofia status is there...
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 able to dial an
extension in closed in a number. So for instants if I dial 120275
extension
PCCW is use for making calls through IP connected through cellphone just
enter the areacode for example
900639274522123
900-prefix
63-areacode
9274522123 - number?
Has anyone has tried it?
Please help me how to connect to it
--
View this message in context:
Hi Giovanni,
I've reported it in Jira. Here's the bug url:
http://jira.freeswitch.org/browse/MODSKYPIAX-35
Thanks,
-Jingwei
On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Hi Jingwel,
thanks for reporting.
Could you please add a Jira issue with as much
Hi Jingwei,
Thanks a lot! I'll take care of as soon as possible.
Btw, before I read the Jira, are you testing in linux?
If you are testing on linux, would you please report how it is
performing under load? I mean, what is the load average with, let say,
10 or 20 or more concurrent Skype call?
I'm currently rewriting the entire thing, it was a commercial app
first, but I'm re-writing it in order to make it open source. It's not
ready yet, as soon as I finish it, I will release it to the public.
Diego
On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruzdarklio...@yahoo.com wrote:
Can you
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to pickup
the call once (to accept the call in X-Lite (or a Snom phone) and to
start the workflow on the web application). I do not want him to accept
the call on the phone and then on the Web app.
Thanks for that info... Can you send me this project if and only if it is
already finished on this email darkl...@yahoo.com? Thanks a lot...
Diego Viola wrote:
I'm currently rewriting the entire thing, it was a commercial app
first, but I'm re-writing it in order to make it open source.
Sure, I'll append to you the result tomorrow.
Regards,
-Jingwei
On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Hi Jingwei,
Thanks a lot! I'll take care of as soon as possible.
Btw, before I read the Jira, are you testing in linux?
If you are testing on
Sure, I will let you know when it's done.
On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruzdarklio...@yahoo.com wrote:
Thanks for that info... Can you send me this project if and only if it is
already finished on this email darkl...@yahoo.com? Thanks a lot...
Diego Viola wrote:
I'm currently
Hi brain,
Are you still looking into this?
I think it must be some error when it register, I manually changed the
contract str in the registration db, immediately it works. After re-
register, stop work again.
Should I report this to jira?
sqlite select contact from sip_registrations
Hello sir,
Do you know how to connect to two freeswitch at a time with different Ip
addresses?
If a user is register on FreeSwitch 1, the user should not have another
account or he/she will not register anymore for Freeswitch 2?
They can call each other...
I already make one but an error
If you can catch brian or me on irc can you provide remote access to
this box and we should be able to fix this pretty quick
Mike
On Jun 16, 2009, at 5:20 AM, seven dujinf...@gmail.com wrote:
Hi brain,
Are you still looking into this?
I think it must be some error when it register, I
The only way I can think to do this today would be to cancel the call
and re send with the intercom headers for a phone that supports it.
It may be possible to send a reinvite with autoanswer headers but I
doubt that would work, all you could do is try making code to do it it
a sipp or
Peter P GMX wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to pickup
the call once (to accept the call in X-Lite (or a Snom phone) and to
start the workflow on the web application). I do not want him to accept
the call on the phone and
what is PCCW? could you please fill in more details what you like to do. to
connect mobile phones w/ FS, the mobile phone has to have SIP feature. pls
search the Wiki for some models.
-nandy
===
LanVox Systems
Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Solved by replacing auto-nat with public ip in public profile
external_sip-ip and extrenal-rtp-ip params.
I believe values for these params used to be taken from vars.xml and so
would have public ips by default - would be nice to document such
changes in README.
paul.d...@gmail.com wrote:
Hello Ray,
I do use event socket and it pushes me a link on the website whenever a
call for this agent comes in.
It's just a matter of visibility. The agent may still finish his old
workflow and is still entering data. When a call comes in then and he
picks up the phone, the data he just entered
Hello Michael,
I want the phone be ringing, just for acoustical feedback reasons.
But what if I
* transfer it to the same user destination again (now with intercom
enabled), will this work?
* transfer it to park and then transfer it to the same destination
again (now with
Can you please put it back to auto-nat and email me the output of
global_getvar from the CLI so I can see what it detected?
/b
On Jun 16, 2009, at 7:18 AM, paul.d...@gmail.com wrote:
Solved by replacing auto-nat with public ip in public profile
external_sip-ip and extrenal-rtp-ip params.
I
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:
Why not just keep the agent off hook.. in park state... then just
playback ringing before you bridge?
/b
On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote:
Hello Michael,
I want the phone be ringing, just for acoustical feedback reasons.
But what if I
* transfer it to the same user
Ok i'll have to se what I can do about reproducing this issue now that
I have more info on how its happening.
/b
On Jun 16, 2009, at 7:40 AM, dujinfang wrote:
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
Did you compiled freeswitch with this command?
./configure --enable-core-odbc-support
makemake installRegards
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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
May this help also: I just tried current Zoiper with TLS. Outbound is
working, inbound not.
Zoiper registeres with the following contact info:
7233213
sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS
When a call comes in, Zoiper rings once and then hangs up. It shows
What's wrong of the contact string? 639(snom) works but 637(zoiper)
doesn't.
user sip:6...@192.168.1.27:5070;rinstance=e1e47e9a22f3e450;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3De1e47e9a22f3e450%3Btransport%3DUDP
seven
The transfer should work but it sounds like offhook agents is what
your really trying to accomplish so I would go with brian's suggestion.
On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote:
Hello Michael,
I want the phone be ringing, just for acoustical feedback
I need sip traces... also can you guys register to my dev box?
dev.bkw.org with default user/pass try 1009 thru 1015 please.
/b
On Jun 16, 2009, at 8:17 AM, Seven Du wrote:
What's wrong of the contact string? 639(snom) works but 637(zoiper)
doesn't.
user
Spread the word!
We have need of some volunteers to assist us with various tasks at ClueCon
this year. As you may know, when putting on a conference there are numerous
little things that require attention. Having several designated volunteers
to handle these tasks will make the conference better
I actually do that with our call center application. For all incoming
calls, our IVR engine parks the call in a virtual extension and plays
back prompts, advertisements, MOH, process digits, etc. When the queue
management finds an available agent, it sends an event to the client
application for
This issue is now fixed in svn. Thanks Seven for access to your box
to troubleshoot.
Mike
On Jun 16, 2009, at 9:17 AM, Seven Du wrote:
What's wrong of the contact string? 639(snom) works but 637(zoiper)
doesn't.
user
Brian,
Thank you for putting me on the right track. I thought I would share
my results so after a bit of trial and error testing I came up with the
follow DP rule, which lives in dialplan/public/my_public_dp.xml.
When an incoming call arrives for DDI 012345678 it is ack'ed with a
180
What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the
real logs from FS's logs? The info below is not nearly detailed enough.
-Original Message-
From: Edmar Cruz [mailto:darklio...@yahoo.com]
Sent: Monday, June 15, 2009 6:44 PM
To:
That should not be the case - I will double check this. My apologies if I
broke it. :-(
Please file a bug on this so I don't forget.
_
From: Yuriy Ivzhenko [mailto:yivzhe...@mksat.net]
Sent: Tuesday, June 09, 2009 1:26 AM
To: freeswitch-users@lists.freeswitch.org
Subject:
This should be fixed in the latest build (thanks MikeJ)
_
From: ram [mailto:talk2...@gmail.com]
Sent: Tuesday, June 09, 2009 12:03 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] mod_niible install problem
Hi
i have downloaded latest SVN
and trying to
I think this question might need to be backed up with some more information.
I recommend you post your relevant configs to pastebin so that we can have a
look. (pastebin.freeswitch.org)
-MC
On Tue, Jun 16, 2009 at 8:17 AM, selva kumar panse...@gmail.com wrote:
Hi,
I've tried configuring the
ClueCon 2009 is only seven weeks away! We are all looking forward to meeting
together in Chicago. To make sure that everything goes as planned we would
like to know how many people will be attending. If you have not already
signed up for ClueCon 2009 please do so. Call 877.742.CLUE and Brian will
It mainly works now by uuid_transfer the following way via event socket.
uuid_setvar unique_id sip_invite_params intercom=true
uuid_setvar unique_id sip_auto_answer true
uuid_transfer unique_id 1000 XML default
so the call is transferred from 1000 to 1000.
What happens:
1) If I
uuid_setvar unique_id sip_invite_params intercom=true should be
unnecessary.
Mike
On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:
It mainly works now by uuid_transfer the following way via event
socket.
uuid_setvar unique_id sip_invite_params intercom=true
uuid_setvar
I'm creating a conferencing product for use in a system with theoretically
several hundred concurrent calls. I'm using FreeSwitch to create this
product, but am not only new to FreeSwitch, but also the entire telecom
industry as well as Open Source projects in general (I'm a recovering BIOS
guy).
Can you describe your networking environment a bit? One thing that can
affect the latency of your voice traffic is your network infrastructure. If
you can isolate FS and some phones on a separate, controlled network then
possibly you can start narrowing it down to other factors.
-MC
On Tue, Jun
The problem comes from the timing of certain phones during the capture of
audio actually clocked slightly faster than what it advertises.
Try the latest trunk with all the defaults in your sip profile as we have
tried to make the defaults deal with this automatically.
On Tue, Jun 16, 2009 at
I have two different network setups, and have seen similar lag on both.
The first is my home testbed. I'm connected to the internet through a home
router and then a cablemodem. The home environment is pretty spare, of
course. 2 machines and a couple of T-mobile cell phones with their SIP
I am not as knowledgeable as the developers that will respond to your question
but I had the same problem as you. Here is what I did to combat the delay:
First off I started everything from scratch. I reinstalled Linux and then I
reinstalled FreeSWITCH by creating .deb packages.
I then
Thanks Michael,
I have disabled it now.
I finally got it to work, (sip_h_Call-Info=sip:$${domain};answer-after=0)
but the behaviour was not as desired, as I didn't manage the phone to
pick up the call on the headset. It will only have the speaker enabled.
So I will have to go a different way
I'm not sure I've got the opportunity to do that at the moment, but I do
appreciate the point of view of a fellow product user. Were you able to
eliminate noticeable lag, or just reduce it to reasonable levels?
I'll try to do something similar when I update to the newest trunk as
Anthony
How much power do I have with DTMF conference controls? The wiki doesn't
have much information on this. For example, one of the things I'd like to do
is take the currently existing lock and unlock actions and merge them
into a lock toggle action. Preferably in XML configuration files. Is this
even
I was able to reduce it considerably. I can't say it is completely gone but I
am very confident the ~.5 second delay I hear is because of the time it takes
my voice to go through the leaps and bounds of the phone company to our server.
I had at least a 3-5 second delay before I experimented
Hi everyone,
Can you please recommend me some GSM gateway? I'm currently looking
for a good one to buy... anyone have experience PORTech GSM gateways?
Are they good?
I also need it to work with FS, I'm also kinda new with VoIP hardware.
Thanks,
Diego
What is the big picture application? Reason I ask is that the FS devs and
community have a lot of experience so if they can see the big picture they
might be able to offer better advice.
-MC
On Tue, Jun 16, 2009 at 2:26 PM, Bradley Brashier bjbrash...@gmail.comwrote:
How much power do I have
Bradley Brashier wrote:
How much power do I have with DTMF conference controls? The wiki
doesn't have much information on this. For example, one of the things
I'd like to do is take the currently existing lock and unlock
actions and merge them into a lock toggle action. Preferably in XML
If you want FS server A to be able to call FS server B, you can set up a
user account in server B's FS directory configs, and then just treat server
B as a normal gateway by adding a gateway definition in server A. That will
allow you to route calls to server B from A; to do the reverse, just
I did a fair amount of research into GSM gateways about 8 months ago. I should first ask what are you looking to do with the gateway?-pete
Original Message
Subject: [Freeswitch-users] Which GSM gateway to buy?
From: Diego Viola diego.vi...@gmail.com
Date: Tue, June 16, 2009
don't forget to read my suggestion too from earlier today =D
On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon jo...@wabashcenter.com wrote:
I was able to reduce it considerably. I can’t say it is completely gone
but I am very confident the ~.5 second delay I hear is because of the time
it takes
Will do, just haven't had the time, yet!
On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
don't forget to read my suggestion too from earlier today =D
On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon jo...@wabashcenter.com wrote:
I was able to reduce it
Get Khomp GSM cars! Ihihihih
They will soon be compatible with FreeSWITCH.
Laterz,
jmesquita
On Tue, Jun 16, 2009 at 6:48 PM, p...@privateconnect.com wrote:
I did a fair amount of research into GSM gateways about 8 months ago. I
should first ask what are you looking to do with the gateway?
I need it for gsm termination, I'd like to start with 8 channels, then 16, etc.
Thanks,
Diego
On Tue, Jun 16, 2009 at 5:48 PM, p...@privateconnect.com wrote:
I did a fair amount of research into GSM gateways about 8 months ago. I
should first ask what are you looking to do with the gateway?
For those that understand Portuguese
http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/
-E
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We are a start-up company btw.
On Tue, Jun 16, 2009 at 6:09 PM, EdPimentledpime...@gmail.com wrote:
For those that understand Portuguese
http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/
-E
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Freeswitch-users
So we can't afford the top and the latest hardware.
On Tue, Jun 16, 2009 at 6:21 PM, Diego Violadiego.vi...@gmail.com wrote:
We are a start-up company btw.
On Tue, Jun 16, 2009 at 6:09 PM, EdPimentledpime...@gmail.com wrote:
For those that understand Portuguese
Hi ,
I have used PORTech single and double channel units on a couple of small
projects with FS and they seem to have worked well in a low volume
application . Have never tried one of the larger channel count ones yet for
high call volumes though so cant verify how they perform, although just
API CALL [global_getvar()] output:
external_ssl_enable=false
external_tls_port=5081
external_sip_port=5080
external_auth_calls=false
internal_ssl_dir=/var/opt/freeswitch/conf/ssl
internal_sip_port=5060
default_provider_contact=5000
default_provider_from_domain=example.com
Hi All,
I have a requirement to delay the audio sent from the calling channel
in a call by a specified delay, much the same as the delay_echo
functionality in the dptools but in a bridged rather than loopback
mode. I cant immediately see a way to achieve this, is this something
I'm missing or
if it's sip, turn on the jiterbuffer
before you answer
set the var jitterbuffer_msec=x
where x is desired number of milliseconds (not too much!)
On Tue, Jun 16, 2009 at 5:57 PM, Steven Brown st...@justfone.com wrote:
Hi All,
I have a requirement to delay the audio sent from the calling
Hmmm is that going to be easier than just modifying the mod_conference
code to allow for a handfull of extra, simple commands? To me, it seems like
for reasons of maintainability, etc, you want as few varied pieces as
possible, in as few languages as possible. Socket scripting doesn't sound
It depends pretty heavily on what you are trying to add function wise. If
it's more in depth using the event socket would allow it to be used on any
FreeSwitch server assuming it caught the dtmf and acted according without
having to modify the core source code/recompile. It might be a bit more
Hello friends.
I've been playing with the mad boss examples. There is an issue I'd like to
see:
For example in MadBoss3:
The first leg added to conference is the loopback/. Then you can add
more users by conference_set_auto_outcall function.
The problem I see is that:
1)
Actually my plan is if FS Server A has an account of 8011105, FS Server B
shouldn't create another directory config. The user most not create an
account 8011105 ON FS Server B. Single account for two servers. When I used
a gateway config, yes its working but it needs a username and password
My
my nibble.conf.xml
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- Information for connecting to your database --
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the
Look at the newly implemented wait-mod conference flag on mod_conference.
This is: http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E
under parameters-conference-flags
jmesquita
On Tue, Jun 16, 2009 at 10:22 PM, Ing. Edwin Villarreal
evi...@chipoly.comwrote:
Hello friends.
Diego, i'have a customer using 3 portech using todo termination on argentina
with asterisk on high volume calls and they are working great.
Best regards.
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I have been using Portech for over two years and they work fine.
-E
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Turn off authentication or use ACL's
/b
On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote:
Is there another way to manage the gateway with the caller id of the
user
not the gateway user id and is there a gateway that doesn't need a
username
and password?
Hello!
I need some fresh ideas about this issue. My gateway is already REGED, but
when REG expires and sofia is trying to renew REG, then it fails to
register.
. 2009-06-16 16:46:39 [ERR] sofia_reg.c:1381
sofia_reg_handle_sip_r_register() chipoly Registration Failed with status
DNS Error
This should be a huge clue... what might be your providers name?
Seems something is missing here or you have the settings wrong.
/b
On Jun 16, 2009, at 9:58 PM, Ing. Edwin Villarreal wrote:
DNS Error [503].
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How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105
list name=fsb default=deny
node type=allow cidr=192.168.0.104/32/
/list
On 192.168.0.4
list name=fsa default=deny
node type=allow cidr=192.168.0.105/32/
/list
Brian West-3 wrote:
Turn off authentication or use
Hello, all. I'm currently playing around with a new install of Freeswitch and
wanted to try out mod_opal. Below are the current SVN builds for opal, ptlib,
and freeswitch. I end up with the following errors when compiling.
making all mod_opal
Compiling mod_opal.cpp...
Compiling mod_opal.cpp
please see MODOPAL-10 on jira.
/b
On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote:
Hello, all. I'm currently playing around with a new install of
Freeswitch and wanted to try out mod_opal. Below are the current
SVN builds for opal, ptlib, and freeswitch. I end up with the
Shouldn't have really changed any behavior at all... What svn rev are
you on?
/b
On Jun 16, 2009, at 5:50 PM, paul.degt wrote:
API CALL [global_getvar()] output:
external_ssl_enable=false
external_tls_port=5081
external_sip_port=5080
external_auth_calls=false
How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105
list name=fsb default=deny
node type=allow cidr=192.168.0.104/32/
/list
On 192.168.0.4
list name=fsa default=deny
node type=allow cidr=192.168.0.105/32/
/list
From: Brian West
Now you have to tell the sofia profile to use that ACL
/b
On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote:
How can i turn off authentication? This is my acl.conf.xml on
192.168.0.105
list name=fsb default=deny
node type=allow cidr=192.168.0.104/32/
/list
On 192.168.0.4
list name=fsa
How can sofia profile can call ACL?
Can you give me an example?
Brian West-3 wrote:
Now you have to tell the sofia profile to use that ACL
/b
On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote:
How can i turn off authentication? This is my acl.conf.xml on
192.168.0.105
list name=fsb
COPY paste fail :)
param name=apply-inbound-acl value=domains/
something like that as per the example.
/b
On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote:
How can sofia profile can call ACL?
Can you give me an example?
Like this?
I put this on external profile
/
/
Brian West-3
If FS A has an account 8011105 does FS B also nid to register 8011105? Yes it
working on a gateway but the username of the gateway was shown on my
softphone and also it nids a password for the gateway... is there an option
to view the caller name and number of the FS A gateway to FS B?
Brian
Its not an error its a warning and you don't have your ACL's
configured correctly. You're trying too hard! :) set auth-
calls=false on the profile.
/b
On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote:
Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on
FS B
hi,
if u need any help, i can always provide that.
Regards,
Bipin
Diego Viola wrote:
Sure, I will let you know when it's done.
On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruzdarklio...@yahoo.com wrote:
Thanks for that info... Can you send me this project if and only if it is
already
13564
Brian West wrote:
Shouldn't have really changed any behavior at all... What svn rev are
you on?
/b
On Jun 16, 2009, at 5:50 PM, paul.degt wrote:
API CALL [global_getvar()] output:
external_ssl_enable=false
external_tls_port=5081
external_sip_port=5080
can you update and try that again?
/b
On Jun 17, 2009, at 12:00 AM, paul.d...@gmail.com wrote:
13564
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Yes its already set to false... What should I do?
list name=fsb default=deny
node type=allow cidr=192.168.0.104/32/
/list
list name=fsa default=deny
node type=allow cidr=192.168.0.105/32/
/list
Brian West-3 wrote:
Its not an error its a warning and you don't have your
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