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hello friends, is there any way to simplify that extensions.conf file? [miprimerejemplo] exten => 20000,1,Dial(SIP/20000,30,Ttm) exten => 20000,2,Hangup exten => 20000,102,Voicemail(20000) exten => 20000,103,Hangup exten => 20100,1,Dial(SIP/20100,30,Ttm) exten => 20100,2,Hangup exten => 20100,102,Voicemail(20100) exten => 20100,103,Hangup exten =>

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

Hi, i'm trying to complîle and install ices2 on my linux slackawre box : After leting make runnins a while , i got these errors ... Would tou help me to solve please ? I have just grabbed ices from cvs. In file included from input.c:29: input.h:34: parse error before "shout_t" input.h:34: warning: no semicolon at end of struct or union input.h:36: parse error before '...

Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>

hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/86e4ad95/attachment.htm

On 27/03/16 07:25, IT Admin wrote: > I ran ldbsearch on my sam.ldb > I searched for CBADC02, CBADC03, and TESTES (all VMs that fail to join > domain), results are below: > > > CBADC02 shows up a few times: > > # record 1906 > dn: > CN=CBADC02\0ADEL:de85228c-f92b-4d5d-9d6a-01c3f915dec9,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configu$ > objectClass: top

...As for what you can control, first, you might try reducing the > expiration from 3600 to 999, or maybe something in the 30-60 range is > better for you. If that works, then raise it from there, but I think an > hour is more than enough. We tried with 99, 60, 986, without setting expiration leting Asterisk using his default value, no changes :( > Or, change network paths; by adding new outbound SIP connection to the > provider from alternate port and/or IP on the PBX/firewall, use VPN, etc. Not a solution, to risky. -- Daniel -------------- next part -------------- An HTML attachme...

On Thu, 2003-01-23 at 06:00, Michael Smith wrote: > Rakotomandimby Mihamina <mrakotom@wanadoo.fr> said: > > > Hi, i'm trying to complîle and install ices2 on my linux slackawre box : > > After leting make runnins a while , i got these errors ... > > Would tou help me to solve please ? > > I have just grabbed ices from cvs. > > You need libshout from cvs as well. > Ok i grabbed libshout, I installed it , now i get ANOTHER error include -c `test -f encode.c || echo './...

I ran ldbsearch on my sam.ldb I searched for CBADC02, CBADC03, and TESTES (all VMs that fail to join domain), results are below: CBADC02 shows up a few times: # record 1906 dn: CN=CBADC02\0ADEL:de85228c-f92b-4d5d-9d6a-01c3f915dec9,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configu$ objectClass: top objectClass: server instanceType: 4 whenCreated: 20160310044543.0Z uSNCreated: 4215

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have a small Sieve script that tries to use vacation that segfaults. The script is the one from http://wiki.dovecot.org/LDA/Sieve. When I change the email address (I changed the local part into "skai"), hence, vacation is skipped, the mail is delivered without any problem. ==== script start require ["fileinto",

Alright... appreciate the info. Gave it a shot. Domain is still up but shares are down because they were hosted on FILER which has now been demoted and is no longer running any samba services. What I did while following the wiki "Transfer/Seize FSMO Roles": 1) logged on to FILER, ran samba-tool fsmo show, verified all 7 roles were owned by FILER. 2) logged on to CBADC01, executed

On Thu, Oct 11, 2018 at 10:55:39AM +0300, Aki Tuomi wrote: > Maybe. Have to see when we can implement it though. It could probably > leverage the min/max_gid setting. Actually that was a great hint. Setting last_valid_gid = 100 in the config and restarting helped. Having a filter-list instead of fixed upper/lower bounds would be more flexible. I guess though that in reality most

...rbisfile * libshout * icecast2 * ices2 <p>> On Thu, 2003-01-23 at 06:00, Michael Smith wrote: > > Rakotomandimby Mihamina <mrakotom@wanadoo.fr> said: > > > > > Hi, i'm trying to complîle and install ices2 on my linux slackawre box : > > > After leting make runnins a while , i got these errors ... > > > Would tou help me to solve please ? > > > I have just grabbed ices from cvs. > > > > You need libshout from cvs as well. > > > Ok i grabbed libshout, > I installed it , > > now i get ANOTHER error...

I have just started using R for my PhD. I am importing my data from Excel via notepad into Word. Unfortunately, my data has many missing values. I have put '.' and this allowed me to import the data into R. However, I now want to interpolate these missing values. Please can someone give me some pointers as to the method/code I could use? Thankyou, Lillian.

hello i have installed Asterisk on a Debian machine by apt-get install asterisk I only want to call a user inside the LAN, what files I have to edit??? sip.conf??? thanks for all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070327/355be0cc/attachment.htm

hello, I have the A400P01 from OpenVox. Is necesary to install all this packages? http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz or just with asterisk and zaptel is enough. thanks a lot -------------- next part -------------- An HTML attachment was

Hello again, I have a little problem, every time I switch on the Asterisk server I must load two modules: modprobe zaptel and modprobe wctdm Is there any way to load there automatically when the server start? I have a Debian Etch. One more cuestion, it's posible to start Asterisk (asterisk -vvvc)as well? What metod do you prefer? "asterisk" or "asterisk -vvvc"? Thanks

Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten => 105,1,Answer exten => 105,2,Background(/home/user/suport) exten => 1,1,Dial(SIP/101,30,Ttm) exten => 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option "1" it doesn't redirect to

Hello everybody! I have a problem recording voices for my Asterisk menu. I used the "Record(/home/lazkano/bienvenido:gsm)" function to record the menu voices, but when I call from outside or from an extension the voice listen so low. is there any software to record my voice properly and convert to gsm format? Someone use an other function for that? Thank a lot to everybody. Enjoy

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a