similar to: AMI - delete voicemail

How to execute some system command from AMI? -- Tomislav Parcina firstname.lastname@email.t-com.hr

I'm using AMI on Asterisk 1.2.15 and I'm having problems with DBPut action. If I execute this: Action: DBPut Family: checkin Key: 316 Val: yes Response: Error Message: Missing action in request I don't put anything in Asterisk DB. If I execute this: Action: DBPut Family: <checkin> Key: <316> Val: <yes> Response: Success Message: Updated database successfully

Does Asterisk supports 302 SIP message - "Moved temporarely"? I have found mail from Olle E. Johansson (April 2006) that Asterisk doesn't support redirects of registrations. Does it support now? -- Tomislav Parcina firstname.lastname@email.t-com.hr

There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something

Hi list! How to send a call directly to voicemail recording? When I put this exten => 313,n,VoiceMail,u221 Or this exten => 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#lama.hr

One simple question. I'm using asterisk 1.2.1, can one agent be defined in more than one group? Example: group=1 ; queue1 agent => 401,401,Tomislav Parcina agent => 402,402,Katarina Ivanisevic agent => 403,403,Sasa Juginovic group=2 ; queue2 agent => 401,401,Tomislav Parcina agent => 402,402,Katarina Ivanisevic agent => 404,404,Marija Bilic agent => 405,405,Ana

As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not "trusting" the asterisk user. And the default behaviour of that is to not allow modification of the "fromstring" and "serveremail". So if you have any idea how to fix that in Gentoo I would really appreciate it. Thanks -----Opprinnelig melding----- Fra:

Hi, I?m a new Dovecot user and using version 2.2.18 on an OpenSuse system. In general it all works quite nicely and clients can connect to Dovecot and manage mails normally. The layout used is maildir. The users are all virtual, i.e. they do not exist on the Linux system. They all log in without any authentication due to a very specialized and internal setup. However, sometimes Dovecot simply

Patch it by hand. Follow this help http://www.asteriskguru.com/tutorials/spandsp.html Hi Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Tomislav Parcina Inviato: venerd? 13 gennaio 2006 12.28 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] RE: RE: Spandsp In article

How can I send recordings, that I have recorded with One Touch Record, to e-mail address that is defined in voicemail.conf? Thank you for your ideas. -- Tomislav Parcina tparcina#lama.hr

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

I have foloved instructions at this web pages http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call contacts from Outlook. Now I have few questions. When I place a call, my phone rings before * tries to dial out. Is it posible that * first dials out, and when other side picks up, at that moment that my phone rings? Another question, when I recive a phone call, can that

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Does Asterisk support Advice of Charge? I was told that my Telco sends me billing signalization that way, and I wonder can I use it? -- Tomislav Parcina ime.prezime@email.t-com.hr

- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is <line button="4"> <featureID>9</featureID> ... For speeddial is <line button="5"> <featureID>2</featureID> <featureLabel>341</featureLabel> <speedDialNumber>341</speedDialNumber> </line>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Sat, 29 Sep 2018, Fady AL HAYALI wrote: > I'm setting up a Postfic and Dovecot with LDAP email server. My users in LDAP is like this: > > dn: uid=firstname,ou=People,dc=domain,dc=com > uid: firstname > uidNumber: 4025 > gidNumber: 4025 > givenName: firstname > objectClass: top > objectClass:

Von unterwegs gesendet > Am 01.10.2018 um 18:27 schrieb Aki Tuomi <aki.tuomi at open-xchange.com>: > > >> On 01 October 2018 at 15:19 Steffen Kaiser <skdovecot at inf.h-brs.de> wrote: >> >> >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >>> On Sat, 29 Sep 2018, Fady AL HAYALI wrote: >>> >>> I'm

On Mon, Oct 01, 2018 at 11:25:48PM +0200, Admin wrote: > > > Von unterwegs gesendet > > > Am 01.10.2018 um 18:27 schrieb Aki Tuomi <aki.tuomi at open-xchange.com>: > > > > > >> On 01 October 2018 at 15:19 Steffen Kaiser <skdovecot at inf.h-brs.de> wrote: > >> > >> > >> -----BEGIN PGP SIGNED MESSAGE----- >