hi,
any1 have any idea how what to sue in dialplan such that calls from a single
id go to a specific gateway only with blind registration enabled, this is
the only major issue im having.
Regards,
Bipin
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View this message in context:
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How do you load balance conference calls? Doesn't all the conference members
have to be on the same freeswitch server?
On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko wrote:
> Brian West pisze:
> > what kind of hardware?
> >
> I made testes on Pentium-M laptop with single core 1,6Hz. I did not write
I restart FS and initiate an incoming call (trunk is registered at the
SIP provider).
This is what I see on the console:
.
.
.
2009-04-02 23:39:16 [DEBUG] mod_event_socket.c:2224
mod_event_socket_runtime() Socket up listening on 0.0.0.0:8021
2009-04-02 23:39:16 [NOTICE] switch_core.c:981
switch_lo
look at the debug log and see what happens?
On Thu, Apr 2, 2009 at 3:34 PM, Peter P GMX wrote:
> My ACL contains:
>
>
>
>
>
> So this should be fine, right? However it doesn't work.
>
> Best regards
> Peter
>
>
> Brian West schrieb:
> > We use the true network ip the invite c
My ACL contains:
So this should be fine, right? However it doesn't work.
Best regards
Peter
Brian West schrieb:
> We use the true network ip the invite came from NOT the one in the sip
> headers. Not very trust worth to do that you think? ;)
>
> So if your ACL is corre
wait for pre4
On Thu, Apr 2, 2009 at 3:04 PM, Ceino wrote:
> Hi all, I have tested it a little bit and it's works well. But when I
> give it the command to stop (...)
> it use about 40 sec to stop (1.0.3 use about 5 sec).
>
> Here is a log over where is hang (looks like a Sofia thread use long
>
We use the true network ip the invite came from NOT the one in the sip
headers. Not very trust worth to do that you think? ;)
So if your ACL is correctly setup to 62.65.128.62 it would let them in
please verify your ACL is correct...
/b
On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote:
He
acl uses the remote addr from the socket connection, not anything from the
sip message.
On Thu, Apr 2, 2009 at 3:07 PM, Peter P GMX wrote:
> Hello,
>
> I am using a SIP account from Netvoip CH. I try to receive inbound call
> from this SIP trunk. I discovered that, when they sent an invite, the
Try updating to SVN trunk... I think we fixed that already.
/b
On Apr 2, 2009, at 3:04 PM, Ceino wrote:
Hi all, I have tested it a little bit and it's works well. But when I
give it the command to stop (...)
it use about 40 sec to stop (1.0.3 use about 5 sec).
Here is a log over where is hang
Brian West pisze:
> what kind of hardware?
>
I made testes on Pentium-M laptop with single core 1,6Hz. I did not write those
results, it was over 100 calls that was handle
good, I was just curios what will happen. Tomorrow I will make real testes. My
production works on 2 core P4 and I have ther
Hi, I have tested it a little bit and it's works well. But when I give
it the command to stop (...)
it use about 40 sec. to stop (1.0.3 use about 5 sec).
Here is a log over where is hang (looks like a Sofia thread use long
time to stop):
--
Hi all, I have tested it a little bit and it's works well. But when I
give it the command to stop (...)
it use about 40 sec to stop (1.0.3 use about 5 sec).
Here is a log over where is hang (looks like a Sofia thread use long
time to stop):
---
Hello,
I am using a SIP account from Netvoip CH. I try to receive inbound call
from this SIP trunk. I discovered that, when they sent an invite, the
IP-Adress of the to: is their own IP address.
There fore ACL doesn't work and FS asks for authorization, which then fails
I receive the following me
what kind of hardware?
/b
On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote:
I did not described it perfectly. I made agents, queues scenarios on
conferences.
This what I tested was for example 100 calls, so it's 200 channels,
and 100 conferences, 2 channels per conference, all are
unmuted. I
Brian West pisze:
>
> On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote:
>
>> I did not think yet about HA nor LB.
>>
>> I tested how FS handles high load. All my calls are placed in
>> mod_conference. When cpu usage gets it's limits then new calls can
>> be placed but sound quality is getting worst
you can't pass it in with -D
you have to actually add
#define SWITCH_DEBUG_RWLOCKS
to the top of switch_core.h
On Thu, Apr 2, 2009 at 11:46 AM, Tamas Cseke wrote:
> Hello,
>
> We originate loopback channels and they end up in calling sofia
> and transfer the call to a fifo.
>
> If we have a he
hehe I emailed it to him off list :)
/b
On Apr 2, 2009, at 1:09 AM, Kristian Kielhofner wrote:
I probably shouldn't be doing this for you, but...
http://bugs.digium.com/view.php?id=14431
;)
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com
__
Thanks for doing some of the legwork on this. BTW, this thread is probably a
bit too technical for the users list - I recommend sending to the dev list.
:)
-MC
On Thu, Apr 2, 2009 at 9:46 AM, Tamas Cseke wrote:
> Hello,
>
> We originate loopback channels and they end up in calling sofia
> and tr
Hello,
We originate loopback channels and they end up in calling sofia
and transfer the call to a fifo.
If we have a heavy call volume loopback-b channels don't hangup properly.
They stay in core.db.
Unfortunetly we can't reproduce it on test boxes but happens every day.
On this box we had to tu
uecon.com
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no, but they all auto-create. You can create a db and set up odbc,
start freeswitch, then dump your db schema. Also, please do not send
confidential emails to the mailing list.
Mike
On Apr 2, 2009, at 11:12 AM, Richard Lamkin wrote:
Are there documents or wiki page [I’ve missed during my
Turn on Multireg too.
/b
On Apr 2, 2009, at 9:38 AM, Bipin Patel wrote:
hi,
will the below work if all the registration that is to be accepted
come
from different public ip addresses, i mean, clients from all ip ranges
and addresses rather than a single ip
Regards,
Bipin
www.xbipin.com
Are there documents or wiki page [I've missed during my searches] that
detail the records and their types that are stored in the various FS
databases; e.g. sofia_reg_.db, core.db ?
Regards
Richard Lamkin
*
This email
Bipin Patel wrote:
> hi,
>
> will the below work if all the registration that is to be accepted come
> from different public ip addresses, i mean, clients from all ip ranges
> and addresses rather than a single ip
>
yeah, that's kinda why its called "blind" ... you don't have to know
where i
sts.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> ___
> Freeswitch-users ma
http://wiki.freeswitch.org/wiki/Mod_conference
On Apr 2, 2009, at 3:29 AM, bmsword wrote:
I want to use another softswitch conference that has been
deployed in freeswitch,How should I do?
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I use the access control list acl.conf.xml to configure that.
Put ip/mask into the domain part of this config file, then it accepts
calls from this ip (eg. ip/32.) or from this subnet (e.g. ip/24).
Best regards
Peter
Leon de Rooij schrieb:
> Hi,
>
> You can blindly accept registrations and
2009/4/2 Anthony Minessale
> Its the buffering and startup of the shout stream taking up the time,
>
> HINT put the shoutcast stream into a local stream with a .loc file and then
> play that in the conference.
>
Ah, that is easy enough! Though I think with icecast doing the
burst_on_connect thi
Its the buffering and startup of the shout stream taking up the time,
HINT put the shoutcast stream into a local stream with a .loc file and then
play that in the conference.
2009/4/1 Rupa Schomaker
> I've setup a conference bridge that has perpetual-sound set to a icecast
> stream. When the
On Thu, Apr 2, 2009 at 8:08 AM, Peter P GMX wrote:
> Wow, this is cool. Fantastic work!
> I tried this immediately. This is also very useful to share data across
> applications.
>
[snip]
>
> I added this info to the wiki.
>
> Best regards
> Peter
>
Thanks for the wiki update -- great to see exa
Hi,
You can blindly accept registrations and / or authentication messages
with these parameters in a sip profile:
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg
regards,
Leon
On Apr 2, 2009, at 3:01 PM, xbipin wrote:
>
> hi,
>
> i wanted to know if there was an
Wow, this is cool. Fantastic work!
I tried this immediately. This is also very useful to share data across
applications.
Here an example how to share data between Freeswitch and a ruby
memcache-client:
On Ruby/Rails I set the namespace e.g. to "freeswitch" for the same
memcached server in environm
hi,
i wanted to know if there was any way to actually accept all registrations
coming towards freeswitch, the normal function is to have all the suerid and
passwords configured, but is there a way to accept all registrations coming
towards a single ip or domain?
Regards,
Bipin
--
View this mes
Update the to the latest. I've added more channel vars:
eg:
after doing:
(not a real number)
I get the following:
variable_lcr_query_digits: [12148267722]
variable_lcr_query_profile: [0]
variable_lcr_query_expanded_digits: [12148267722, 1214826772, 121482677,
12148267, 1214826, 12148
On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote:
I did not think yet about HA nor LB.
I tested how FS handles high load. All my calls are placed in
mod_conference. When cpu usage gets it's limits then new calls can
be placed but sound quality is getting worst with every next call.
When calls
Can you describe the reasoning behind needing to route option packets
via the dialplan?
/b
On Apr 2, 2009, at 3:58 AM, Rajagopal, Sridhar (Sridhar) wrote:
Hi all,
Whenever freeswitch recieves INVITE SIP packet, It forwards the
packet based on the dial plan. I want to use the same dial pla
hi,all
I want to use another softswitch conference that has been deployed in
freeswitch,How should I do?
andy
2009-04-02
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Thanks for taking the time to help me.
Giovanni, I assume you turn comfort noise off by setting it to 0 which I've now
done. How can I tell which codecs I'm using in conference and how would I
change them. The sound is ok on everything else.
Thanks again
From: stormin.nor...@hotmail.co.uk
T
hi,all
I want to use another softswitch conference that has been deployed in
freeswitch,How should I do?
thanks!
andy
2009-04-02
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Hi,
I want to use module lcr to find a best route and his rate , then make a call
and bill on that rate with nibblebill module.
lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" variable
for new channel.
To use nibblebill i need to set "nibble_rate" = "lcr_rate".
What is bes
Ashley van Gerven pisze:
> Hi,
>
> I can't find much info on setting up a redundant or heavy load
> FreeSwitch implementation. Are there any
> links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ?
>
> I imagine the entry level solution is to have two FS boxes configured
> iden
Hi all,
Whenever freeswitch recieves INVITE SIP packet, It forwards the packet based on
the dial plan. I want to use the same dial plan to forward incoming OPTIONS
packet. Please let me know If I need to write my own code for that or is there
any such option in our code base.
Regards,
Sridhar
Hi Ashley,
One easy solution is to use a SIP proxy (opensips/kamailio/...) in front
of FS boxes to load balance the charge between boxes.
FS already has mechanisms to limit number of calls per boxes ( in
switch.conf.xml: max-sessions and sessions-per-second ),
that you can couple to load_ba
Hi all,
background:
mod_skypiax is Skype compatible endpoint that allows incoming and
outbound calls to/from the Skype network and SkypeOut service. It's
seen by FS like other endpoints, so you can originate from sofia,
bridge to skypiax, and connect the call to a landline number via
SkypeOut serv
Hi,
I can't find much info on setting up a redundant or heavy load FreeSwitch
implementation. Are there any
links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ?
I imagine the entry level solution is to have two FS boxes configured
identitcally, with
redundant SBC software (re
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