similar to: 302 Moved temporarely

How to execute some system command from AMI? -- 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

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

Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) -------- VoIP Provider ------- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also

Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks

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

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

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

How can I delete voice mails (all, new and old) from AMI? I thought that I could use Action Command, but there is no command to delete voicemail. So I figure it out to use system command and execute rm /var/spool/asterisk/voicemail/default/100/INBOX/* and rm /var/spool/asterisk/voicemail/default/100/tmp* but I don't know how to do that from AMI. Any suggestions how to do this are welcome.

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

Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

Hi, 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: person objectClass: posixAccount objectClass: shadowAccount objectClass: organizationalPerson objectClass:

> 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 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 > >

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

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

Hi together, I installed samba 4 on a server with Debian-Wheezy-OS. Everything seems to work fine. I can add a user with RSAT. The user is added to the AD, the profile is stored in the server-profile-path and homepath can be accessed. I have to import about 400 datasets to the AD. Therefore I want to use the commandlinetool dsadd. With this tool I can import datasets, which I can see with RSAT.

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----- >

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

What is command when I wona to list something page by page in * CLI? Something that works like |less or |more. Have a nice day! -- Tomislav Parcina name.surname@email.t-com.hr