similar to: some sip questions

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Please I need help, I don't know why,almost every time I dial on my dect phones, the dialtone doesn't go off and * doesn't recognise anything!!!! I'm using two dlink voip gateways, MGCP: DL102s. Any ideas? thanks in advance michelle matis ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis!

Hi all, i want to use authentication in h323.conf. How can i use it ? In the h323.conf is writed : "If you wish to use Authentication you need to set the appropriate auth keyword above" Where and how i have to set this keyword ??? I've tried "auth=yes",but it does not work. Thanks for Help, Thomas.

Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My

<P>Hi everybody</P> <P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P> <P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't

HI, can somebody tell me how and where must I put the SIP register line? I think is in [general] section of the sip.conf and that I have to put: register => user:password@host:port/localextension but, user and password of the SIP gateway? Because I'm trying this and doesn't work... thanks a lot in advanced michelle ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

Howdy, I''m attempting my first rails test & I could use some help... I have the test posted at http://rafb.net/paste/results/YjWkIh45.html... when I run it, the test fails, but the db table is never restored & it seems that the fixtures file was never loaded... Any ideas as to what I''m doing wrong? Thanks! -- Posted via http://www.ruby-forum.com/.

I'm using * as a Call Agent for two DL102S but I have some problems, like the tones not being sending from the phone to the *. I have not changed the configuration of the DL, except the IP and the Notify Entity (*). Must I change another thing in * or in the device? Thanks very much michelle <br><br> <b>Tu cuenta de correo gratuita <a

Hi all, I'm making a small voip network with asterisk and Dlink dg102s gateways (MGCP). Dlink gateways work with g.711, g731 and g729 codecs. The questions: do i need a g729 codec license for asterisk for dlink use g.729 or just configure asterisk to inform gateways to utilize then ? Anyone have a .conf(s) example(s) to send-me ? Thank's Alexandre da Rosa --- Este e-mail foi

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

On Sat, May 04, 2019 at 06:42:47PM +0200, Thomas K?nig wrote: > > > - figure out Fortran2003 specification for C/Fortran interoperability > > -- this _sounds_ like the right solution, but I don't think many > > understand how to use it and what is implied (in particular, will > > it require making changes to LAPACK itself?) > > That would actually be fairly

I've tried to configure samba to lock files bewteen windows and linux but i couldn't i've read a lot of messages here, but trere is no one that have something about the file smb.conf. i have this in my global secction but i doesn't work [global] workgroup =3D GMC create mask =3D 0777 os level =3D 16 directory mask =3D 0777 hosts allow =3D

Without getting too deep into my app I''m wondering if anyone has any clues what might give rise to the error posted below. It occurs when I run my unit tests. class Actor < AR:Base belongs_to :court_case end class CaseCause < AR:Base belongs_to :court_case end class CourtCase < AR:Base has_many :actors has_many :case_causes ... end def my_test_method ...

Hi, gfortran trunk and 9-branch now have an option to automatically generate C prototypes for old-style F77 procedures. I just did for a in *.f; do gfortran -fsyntax-only -fc-prototypes-external $a > ${a%.f}.h; done in the src/modules/lapack directory. This generates header files which contain prototypes like int ilaenv_ (int *ispec, char *name, char *opts, int *n1, int *n2, int *n3,

Hi all. I am using the library Microsoft RTC to connect a windows software with * in a local network via SIP. In my sip.conf I have defined a 'friend' for this software with their username and their password. I don't want that nobody can be connected without the password. When I call to * it says: "Proxy Authentication Required" in the SIP debug. Please somebody can