similar to: Accessing an account from more than one phone

Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'll create another user. Thanks Luca Bertoncello (lucabert at

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,Rc)

On some SIP phones it is possible to turn off the missed call notifications, but I am not aware of a way to do the same on any cell phones. On 5 Jun 2015 07:29, "Mehmet Avcioglu" <mehmet at activecom.net> wrote: > > There is no signal that is sent to display a missed call. Your cell phone > does that. If it rings and is not answered it counts that as a miss. The > only

Shalom to you too So that's the way to go ? ????? ?????? ? ???: Luca Bertoncello ????: ??? ????, 5 ????? 2015 09:51 ??: Asterisk Users Mailing List - Non-Commercial Discussion ??? ?: Asterisk Users Mailing List - Non-Commercial Discussion ????: Re: [asterisk-users] ?????: Accessing an account from more than one phone Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel!

Zitat von Remko Lodder <remko at freebsd.org>: Hi Remko, > Emails can only be read if they are authenticated / authorized in > someway to access the store. That means you might need to share the > info@ credentials with the other > people so that they can read it over imap or webmail etc. That is self-evident and it is not a problem. I can't understand what you

Hi again! About my previous E-Mail... I though about it and I think, that maybe I'm just very stupid... Since I called an INTERNAL number, Asterisk tried to call it. I tried right now to call an EXTERNAL number (using my context [myproxy]) and the behavior is NOT the same... Not 100% correct, but it tries the right way... Now my problem is to check in my dialplan if the peer, that

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a

Zitat von Aki Tuomi <aki.tuomi at dovecot.fi>: Hello Aki, > Or you can use shared mailboxes... > https://wiki.dovecot.org/SharedMailboxes/Shared Understand I right, that in this case, I __NEED__ all users to have an account on the server? Thanks Luca Bertoncello (lucabert at lucabert.de)

Zitat von Rowland penny <rpenny at samba.org>: > This isn't a bug, it is a feature, winbind no longer returns > anything from AD for 'getent passwd', but it will return the info > from 'getent passwd USERNAME' Hi Rowland, thank you for your answer... I think, this is not so fine, but if this is not a bug, but a feature, I cannot do other but accept it...

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Monday 06 Jul 2015, Luca Bertoncello wrote: >> Well, but for voice quality, which codec is better? >> alaw or gsm? > > A-law is better for voice quality (sorry, thought my original > explanation was > obvious). But note that if the destination is a mobile phone, GSM will be > used anyway, at

Hi list! Since I think, I have a problem in my dialplan, how can I debug it? It would be very useful a command in Asterisk CLI to ask Asterisk what it would do if the number X call the number Y. Something like "exim -bt", if someone here know the SMTP-daemon Exim... Is there such an option in Asterisk? Thanks Luca Bertoncello (lucabert at lucabert.de)

Zitat von Sébastien Le Ray <sebastien-samba at orniz.org>: >> The very strange thing is, that gpupdate tries to copy somethings >> from \\cch.intra\sysvol and not from \\dc1\sysvol... >> There a no server with name cch.intra, this is just the Realm... > > Thats expected. your.realm should resolve to all your DC in a > round-robin fashion. OK, I didn't

Zitat von "L.P.H. van Belle" <belle at bazuin.nl>: > Yes, this is a bug somewhere, and my guess is its related to the > precompiles debian/ubuntu packages. How beatiful... :( > Try > getent passwd username > id username > wbinfo -g > do these work? They work... Thanks Luca Bertoncello (lucabert at lucabert.de)

Zitat von Luca Bertoncello <lucabert at lucabert.de>: > I removed the double browseable, but the situation didn't change... > > But I have notice something stranger: the problem just happen with two users > in the "Domain Admins"-group. > With another user, not in this group, new created files and directories have > the right owner... Well, I noticed right

Zitat von jg <webaccounts173 at jgoettgens.de>: > Yes, it is called "core set verbose 42", the other options is "core > set debug 42". Enjoy the show! OK, thanks, but with this option I can just debug what happens if I call an extension right now... I'd like to have a command to ask Asterisk how it will handle a call... > Once you are more familiar

Hi again! With Dovecot 2.2.9 authenticating against the Active Directory I have following problem: - if I login using LOGIN, PLAIN or CRAM, the username is REALM\login (in my case: CCH\lucabert) - if I login using GSSAPI, the username is just login (in my case: lucabert) this makes the access to the mailbox very difficult, since I don't what can I write in mail_location... If I login with

Hi list! I have Dovecot 1.2.9 from Ubuntu repositories on my server. Now I installed Horde and it give me sometimes the error "Mailbox does not support mod-sequences". Well, I must say, that I didn't know these mod-sequences, but I can' know all... Well, the question now is: how can I enable the support of the mod-sequences on the mailboxes of my Server? I searched in the

"Brian ::" <bc at iptel.co> schrieb: > sip trace? Could you please explain? I'm not a VoIP-expert... Thanks Luca Bertoncello (lucabert at lucabert.de)