similar to: Voicemail: saycid without prefix

John Kiniston <johnkiniston at gmail.com> schrieb: > The easiest solution may be to strip the leading zero's off your caller ID > before your caller enters the Voicemail app to leave you a message. > > > ExecIf(REGEX("^[0][0]." > ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) Thanks! I already had this idea and implemented it. It works...

The easiest solution may be to strip the leading zero's off your caller ID before your caller enters the Voicemail app to leave you a message. ExecIf(REGEX("^[0][0]." ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Hi list! > > Yesterday I set up a voicemail on my

On Monday 06 Jul 2015, Luca Bertoncello wrote: > John Kiniston <johnkiniston at gmail.com> schrieb: > > The easiest solution may be to strip the leading zero's off your caller > > ID before your caller enters the Voicemail app to leave you a message. > > > > > > ExecIf(REGEX("^[0][0]." > >

Hi list! I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly. I configured a voicemail and I receive an E-Mail with some information about the call. Again, wonderful! Now my wish: I'd like to have Asterisk to search the caller in a list file and send me the name corresponding to the number in the E-Mail of voicemail. Is it possible? I currently use ${VM_CALLERID} in

Hi again! I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too... I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME

Hi list! My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesn't... I get this error: [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Oct 17 17:01:29] WARNING[14700]: file.c:957

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

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)

Am 23.06.2020 09:28, schrieb Marek Greško: Hi > if you need clampmss then it is highly probable there is a PMTU > discovery problem. The clampmss does not work for UDP. Is there a way to check if I have this problem? > I probably counted the size incorrectly. So you are able to ping with > size 1464 and not with 1466. How about trying same ping sizes from the > internet towards

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 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

Am 23.06.2020 10:07, schrieb Marek Greško: Hi > this is a correct response: > > From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set > (mtu = 1492) > > So PMTU discovery is working. No problem here. You got correct message > to lower the packet size from 62.156.246.57. This is probably the last > hop before your site. No, the last hop is 62.156.246.65:

Patrick Laimbock <patrick at laimbock.com> schrieb: > On 12/30/15 12:24, Luca Bertoncello wrote: > > Ishfaq Malik <ish at pack-net.co.uk> schrieb: > > > >> Do you have a link to the user guide for your exact phone model? > > > > Unfortunately not... > > I have a Thomson ST2022, but I can just find in Internet manual for the > > ST2030...

Hi! I have a Asterisk installation to manage my phones at home (provider is Deutsche Telekom). It works, but very often the voice is "broken"... Yesterday during a call it was very difficult to understand what my partner sayd... It can NOT be a problem of other downloads/uploads, since in that moment there were no ones... I already had the problem in the past, solved it enabling the

Rowland penny <rpenny at samba.org> schrieb: > Hmm, not what I thought, but if that is the case, two more questions: > > Why are you using the depreciated ntvfs ? So I found in an HowTo (I don't have the address anymore...) > If you joined as a DC, why do you have this in smb.conf: server role = > MEMBER SERVER What have I to use? Thanks Luca Bertoncello (lucabert

Hi list! I configured Asterisk to forward the incoming call for a number to both phones. I wrote that: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,R) of course it works... Now the problem is, that when a phone get the call, on the other phone I get "1 missed call"... Is it possible to avoid that and signaling the other phone, that the call was

Hi list! I already had the problem last year, then it would be solved (surely from some technician by Deutsche Telekom on their servers), and now I have the problem again (but I didn't changed my Asterisk configuration). The problem: after 15 minutes will the call dropped, but only if the call is to another nation! If I just call another phone in Germany, I can speak longer than 15

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

Am 23.06.2020 08:43, schrieb Luca Bertoncello: And another thing, I discovered right now... > Could you suggest me something to restrict the problem? > Currently, I think the problem can be: > > 1) on Asterisk > 2) on my Gateway/Firewall A couple of years ago I added this entry in my firewall: /sbin/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS

Hi list! I have Samba 4.3.8 installed from Ubuntu-Repositories on a Ubuntu 14.04 server. It worked fine as AD-Controller, then today I performed an update, since Ubuntu sayd, that new update are available. Well, maybe I'll NOT do that anymore, since I have many problem, now... I solved almost all of them, but this problem unfortunately not... I use nss and winbind. The users are present