Hi all,
I am running freeswitch on powerpc processor. I see memory being allocated for
each subsequent REGISTER requests coming to freeswitch. But not all the memory
allocated is not freed. If I run the code for two days the system is running
out of memory (RAM available to me is very less).
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Peter,
hmmm well, I had the same idea and I tested it! But ... you have to
make sure that the english grammar/acousticModel is able to cover all
german noises. E.g. I tried to detect Burke, Jan and Gerd. I was
able to map Burke successfully in
If this is Linux, there's nothing wrong with it using most of the
memory, if it starts to use the swap, then there might be an issue.
Utilizing the memory does not mean there is a memory leak
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
Hi Chris, sorry for the late reply. Have been quite busy last few days.
I had shifted 888 from default.xml to public.xml and the dialplan is simply
having an echo action now. I've turned on dl_debug but unfortunately didn't
find anything useful. Logs are attached for your reference.
I don't
Which twinkle version are you using? I use Twinkle 1.0.1 and my SIP
register looks as follows. As you can see, the contact header is there.
U 127.0.0.1:5062 - 127.0.0.1:5060
REGISTER sip:127.0.0.1 SIP/2.0.
Via: SIP/2.0/UDP 192.168.178.146:5062;rport;branch=z9hG4bKtvnvzdwy.
Max-Forwards: 70.
To:
Jingwei, can you show your console log when somebody is calling you from
gtalk client? Will it really hit 888 in your dialplan?
Thanks,
Chris
On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, sorry for the late reply. Have been quite busy last few days.
I
Hi,
I'm fairly sure my problem lies with my voip provider VoipTalk but wonder if
you could help me understand a couple of things. My config is very simple,
I'm using freeswitch to accept incoming calls via a voip gateway and record
messages. Here's the problem:
- When freeswitch starts the
Its not a bug... its just something we do not support in FreeSWITCH
yet... Register with no contact is a fetch operation.
/b
On Jul 13, 2009, at 4:30 AM, Peter P GMX wrote:
Which twinkle version are you using? I use Twinkle 1.0.1 and my SIP
register looks as follows. As you can see, the
Dialogic is coming to ClueCon this year (this aug 4th) and they are
sponsoring the conference.
I can discuss the possibility of supporting their cards at that time.
On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun timuc...@gmail.com wrote:
We have some older dialogic cards (D300 series E1 cards)
Which revision of FreeSWITCH are you using? Several memory leaks have been
fixed since the last formal release. One specifically in REGISTER.
You should probably try SVN trunk or the latest pre-release of 1.0.4
On Mon, Jul 13, 2009 at 2:07 AM, Rajagopal, Sridhar (Sridhar)
Being now a mashup of several CTI companies, there are now a number of
disparate things called Dialogic cards. Some, like the cards previously
known as Prince. er, Eicon are perfectly supportable. The old
Dialogic cards, like the D300 series, are not duplex to and from the
host. They are only
Tim Uckun wrote:
We have some older dialogic cards (D300 series E1 cards) and I am
wondering if freeswitch can support these cards.
Oh, I like the easy questions. No. It lacks the hardware features to do
anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or
anything else
ok,
or we could ask Steve I guess. =D
On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood ste...@coppice.orgwrote:
Tim Uckun wrote:
We have some older dialogic cards (D300 series E1 cards) and I am
wondering if freeswitch can support these cards.
Oh, I like the easy questions. No. It
Maybe Dialogic would add support for thin blades, currently used for HMP
(DNI series).
Val.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Steve
Underwood
Sent: Monday, July 13, 2009 11:40 AM
To:
Are they ignoring the options packet we send them or are they maybe getting
lost behind NAT?
we send an OPTIONS and even if we get a error back we consider that a
successful reply.
We did have a patch into SVN very recently to correct a problem with OPTIONS
ping in a NAT situation.
Maybe try
Got it! Thanks very much for that clarification.
Phil
On Sat, Jul 11, 2009 at 1:43 PM, Jeff Lenk jl...@frontiernet.net wrote:
Hi,
The base dll (FreeSWITCH.Managed.dll) is loaded from /mod - additional
managed dlls are loaded from /mod/managed. This is designed to allow your
dll's to be
Hi,
I purchased a block of 10 did numbers. Base number is 01234/56789. The numbers
themselves range from 01234/567890 to 01234/567899
What works? Well, I can dial in to a hardcoded 01234/56789 which belongs to
user 1000.
I can´t dial out.
The main problem is, that I do not know how I can
If you're on SVN trunk you no longer have to use a double nat
profile.You can set the local-network-acl and ext-[rtp|sip]-ip
settings correctly.
/b
On Jul 13, 2009, at 12:08 PM, Kozak Vladimir wrote:
Please, look at these logs.
Fist invite witchout SDP (from
/attachments/20090713/25e82735/attachment-0001.html
--
Message: 2
Date: Mon, 13 Jul 2009 10:20:53 -0500
From: Anthony Minessale anthony.miness...@gmail.com
Subject: Re: [Freeswitch-users] Help Regarding memory leak with
freeswitch
To: freeswitch-users
/attachments/20090713/25e82735/attachment-0001.html
--
Message: 2
Date: Mon, 13 Jul 2009 10:20:53 -0500
From: Anthony Minessale anthony.miness...@gmail.com
Subject: Re: [Freeswitch-users] Help Regarding memory leak with
freeswitch
To: freeswitch-users
Hello,
I have been playing around with gateway settings today and noticed
something that I wasn't sure if it was a bug or if its just the way it
works.
When I have from-domain set in my gateway config it correctly uses the
configured from domain. If I then set caller-id-in-from to true
Can you collect up sip traces and open a jira please.
/b
On Jul 13, 2009, at 2:04 PM, Dale wrote:
Hello,
I have been playing around with gateway settings today and noticed
something that I wasn't sure if it was a bug or if its just the way it
works.
When I have from-domain set in my
helpless
On Fri, Jul 10, 2009 at 6:44 PM, Jens Vegeby j...@vegeby.nu wrote:
You might wanna write what you need help with :)
On 7/10/09, Ney Frota n...@frota.net wrote:
Help
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On Tue, Jul 14, 2009 at 3:39 AM, Steve Underwoodste...@coppice.org wrote:
Tim Uckun wrote:
We have some older dialogic cards (D300 series E1 cards) and I am
wondering if freeswitch can support these cards.
Oh, I like the easy questions. No. It lacks the hardware features to do
anything
What are the recommended cards to be used with freeswitch?
Sangoma cards and Zaptel/DAHDI compatible cards work well.
-MC
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Using mod_event_socket in outbound mode, is there any to prevent a call from
being disconnected when the outbound socket is closed ? I would like to handle
the initial inbound call using outbound but after the disposition of the call
is determined, close the socket and have that call managed
On Sun, Jul 12, 2009 at 1:49 AM, Eli Hayun eliha...@gmail.com wrote:
In the Perl example I found:
How to access request parameters and how to return data
You have two hashes that are populated for you by freeswitch. Those
hashes are:
* %XML_REQUEST
* %XML_DATA
I want to use
I don't know if this will work for you but I just tested this scenario with
uuid_park. After parking the call I disconnected the socket and the call
continued. I did the same thing with uuid_transfer. After the transfer I
disconnected the socket and the call continued.
How are you handling the
I noticed it in testing last night using net cat. I killed netcat and the
inbound call was disconnected, I'll try your suggestions tonight. Thanks for
the reply,
eric
From: freeswitch-users-boun...@lists.freeswitch.org
Hi,
I tend to believe that we already had this working. Here is my origination
string:
{effective_caller_id_name=Paul
Gascogne,effective_caller_id_number=16478343812}sofia/gateway/sip.gafachi.com/164783486421
The caller number is not being passed to the destination. Is there something
i'm
You need to escape the spaces with \s in the caller id name.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 13-Jul-09, at 10:34 PM, Klaus Teller wrote:
Hi,
I tend to believe that we already had this working. Here is my
It doesn't seem to work though. I tried removing the space completely as well
as removing the caller name parameter.
Original-Nachricht
Datum: Mon, 13 Jul 2009 22:36:37 -0400
Von: Mathieu Rene mrene_li...@avgs.ca
An: freeswitch-users@lists.freeswitch.org
Betreff: Re:
Oh you're using effective_caller_id_number, those vars are only
checked when an a-leg exists.
Use origination_caller_id_number and origination_caller_id_name.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 13-Jul-09, at
Klaus,
Use ngrep and see if the From / RPID headers are correct in the SIP
message. This will let you know if FS is doing the correct thing.
SDR
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Thanks folks. Indeed i had to use origination_caller_id_number.
Cheers,
Klaus.
Original-Nachricht
Datum: Mon, 13 Jul 2009 22:47:18 -0400
Von: Mathieu Rene mrene_li...@avgs.ca
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Gafachi no passing caller
Hi Chris, I've attached the console logs for your reference. It really hits
888 in the dialplan and the external call can hear the echo without any
problem.
One thing attracts me is how the ip addresses are translated. Here's the
working external example:
ses:candidate address=*192.168.2.150*
2009/7/14 Michael Collins m...@freeswitch.org:
What phone number do you call back? I mean, how do you know what the
customer's number is? Do you go by the caller id number?
yes callback to caller id
-MC
On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost d...@tel.co.th wrote:
Dear sir,
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