similar to: Debugging dialplan

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

jg <webaccounts173 at jgoettgens.de> schrieb: > It doesn't really depend on your sip.conf and Asterisk. Your gateway/router > will be the major problem. My summer project will be to look at session Are you sure? Right now I'm using an italian SIP-Provider (Messagenet), configured in my sip.conf and I can receive calls without any problem... So, I don't think, I have to

Hi list! I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with

Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: Hi Sebastian > Brian suggests to check the SIP traces. You can either enable SIP > debugging in Asterisk like so: > > sip set debug on > > Or you could run tcpdump and capture the SIP traffic. > > The first option is probably the easiest. I tried with sip set debug 42 sip set verbose 42 The result was

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

"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)

Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: > I don't remember seeing anything looking like a SIP trace in your first > mail. Try > > sip set debug on > > instead of > > sip set debug 42 > > I don't think there's a sip debugging level like 42 in Asterisk. You can > either switch it on or off. Is it not this:

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 list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format

Hi and happy new year! My question: - two extensions: 1111 and 2222 - an active call on 1111 - incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222 I know how can I forward an incoming call to more than an extension, but I have no idea how can I get the information, that 1111 has already an active call... I think, I need something like: exten =>

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

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

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

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

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)

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)