Dear All,
How to create another user agent like 1000 to 1919 in internal
profile.
Please provide some steps to do it..
Thanks in Advance,
Velusamy
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2009/7/14 Michael Collins :
> 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 wrote:
>>
>> Dear sir,
>> I want to user d
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:
*(external party's local addr)*
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
> An: freeswitch-users@lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Gafachi no passing caller number
> Oh you
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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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 1
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
> An: freeswitch-users@lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Gaf
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
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 mi
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
[freeswitch-users-boun...@lists.fre
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 cal
On Sun, Jul 12, 2009 at 1:49 AM, Eli Hayun 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 LUA to
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?
-MC
On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost wrote:
> Dear sir,
> I want to user dialplan callback to customer. is posible to
> to this is dialplan XML
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 us
> What are the recommended cards to be used with freeswitch?
>
Sangoma cards and Zaptel/DAHDI compatible cards work well.
-MC
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On Tue, Jul 14, 2009 at 3:39 AM, Steve Underwood 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 useful with Fre
Hi guys!
I'm a novice in VoIP world, and may be missing some important concepts, but
recently I've faced a problem with client softphone residing behind a
port-restricted NAT and a public FS server and can't find an explanation on
why it is happening and how to escape it.
Okey, the problem is as
helpless
On Fri, Jul 10, 2009 at 6:44 PM, Jens Vegeby wrote:
> You might wanna write what you need help with :)
>
> On 7/10/09, Ney Frota wrote:
> > Help
> >
> > ___
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> > http:
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
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 the
WITCH Developer Conference
> sip:8...@conference.freeswitch.org <
> sip%3a...@conference.freeswitch.org
> >
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
>
> >
> pstn:213-799-1400
> -- next part --
conf+...@conference.freeswitch.org
pstn:213-799-1400
-- next part --
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Message: 2
Date: Mon, 13 Ju
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 1...@uat.agent.starpoundte
You need to define variables within the user's entry, you can then re-
use those in the dialplan to route the call using the proper line.
If you are using a single trunk you can set the
effective_caller_id_number to the number you want to call from and
it'll set the callerid accordignly.
PS:
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 as
Got it! Thanks very much for that clarification.
Phil
On Sat, Jul 11, 2009 at 1:43 PM, Jeff Lenk 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 built and maintain
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 late
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:
ok,
or we could ask Steve I guess. =D
On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood 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 hard
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
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 d
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) <
sridh...@alcatel-l
and if you go the uuid_setvar route you do this:
api uuid_setvar joe=out_to_lunch
On Sun, Jul 12, 2009 at 6:00 PM, Brian West wrote:
> call-command: execute
> execute-app-name: set
> execute-app-arg: fred=out_to_lunch
>
>
> On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote:
>
> As in
>
> call
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 wrote:
> We have some older dialogic cards (D300 series E1 cards) and I am
> wondering
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
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 gate
Hello Helmut,
the 3 mentioned words are already part of the englisch standard
dictionary, so maybe this causes the problem? You may test with words
which are outside of the standard grammar files or delete the original ones?
So far I have no other documentation available. This part of
PocketSphinx
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 wrote:
> Hi Chris, sorry for the late reply. Have been quite busy last few days.
>
> I had shifted 888 from d
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: "
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 think
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
[mai
-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 successful
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).
T
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