similar to: Allowing calls - maybe I'm just stupid...

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Thursday 11 Jun 2015, Luca Bertoncello wrote: >> Now my problem is to check in my dialplan if the peer, that originate >> the call, is reachable, and if not, to give an error... >> >> Is there any function to know if the peer is reachable? > > The peer that *originated* the call *must* be

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Zitat von Guido Falsi <mad at madpilot.net>: > So, trying to bind authentication to originate calls to registrations is > conceptually wrong in the SIP world. Maybe you can do that but that's > not the way the protocols have been engineered to work. Hi Guido, thanks for your answer. Well, I decided to do that, since I have my Asterisk reachable from Internet just for my

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

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

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > Yes. You should definitely be using A-law for calls to the Outside World. Well, I wanted to change these settings, but I'm not sure, where I have to do that... I think in the users.conf, but I think, the "allow" keywords is for the network... How can I change this setting? Thanks Luca Bertoncello (lucabert

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: Hi, > GSM is the native codec used for calls to mobile phones; it uses lossy > compression to achieve a low bit rate. > > A-law is the native codec used by physical exchanges on the land line network > (PSTN and ISDN). It is non-lossy. It works by arranging the "steps" closer > together near the zero

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

2015-06-08 22:35 GMT+02:00 D'Arcy J.M. Cain <darcy at vex.net>: > On Mon, 8 Jun 2015 22:24:33 +0200 > Luca Bertoncello <lucabert at lucabert.de> wrote: > > Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > > > Basically, they are hoping that you are running the equivalent of a > > > mail server open relay. They are trying to use you

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...

Hi list! I noticed that when the phone of my wife calls the gsm codec will be used, but if someone calls the phone, alaw will be used: 00493511111111 calls 00493512222222: OpenWrt*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No

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 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 "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)

Hi list! I'm working hard to securing my Asterisk... Now I deleted all possibility to access the node as "anonymous" and every call through the proxy will be checked (just known peers are allowed to use it). Furthermore, I restricted the registration of my home phones to the Network I reserved for them and I changed the port on my Firewall, so that I don't use 5060 anymore. Now

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>: > Did you try a samba-tool ntacl sysvolreset on the DC? (actually… > that almost never fixed anything in my case but why not) I just tried to run "samba_dnsupdate --verbose --all-names" on the DC and I got many "Failed nsupdate: 2" and at the end "Failed update of 21 entries". Attached is