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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

...s: server > instanceType: 4 > whenCreated: 20160310044543.0Z > uSNCreated: 4215 > objectGUID: de85228c-f92b-4d5d-9d6a-01c3f915dec9 > systemFlags: 1375731712 > dNSHostName: cbadc02.cb.cliffbells.com > cn:: Q0JBREMwMgpERUw6ZGU4NTIyOGMtZjkyYi00ZDVkLTlkNmEtMDFjM2Y5MTVkZWM5 > isDeleted: TRUE > name:: Q0JBREMwMgpERUw6ZGU4NTIyOGMtZjkyYi00ZDVkLTlkNmEtMDFjM2Y5MTVkZWM5 > lastKnownParent: > CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configurati > on,DC=cb,DC=cliffbells,DC=com > isRecycled: TRUE > whenChanged: 20160319092438.0Z > uSNChanged: 4261 > dist...

Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to

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

...,CN=Configu$ objectClass: top objectClass: server instanceType: 4 whenCreated: 20160310044543.0Z uSNCreated: 4215 objectGUID: de85228c-f92b-4d5d-9d6a-01c3f915dec9 systemFlags: 1375731712 dNSHostName: cbadc02.cb.cliffbells.com cn:: Q0JBREMwMgpERUw6ZGU4NTIyOGMtZjkyYi00ZDVkLTlkNmEtMDFjM2Y5MTVkZWM5 isDeleted: TRUE name:: Q0JBREMwMgpERUw6ZGU4NTIyOGMtZjkyYi00ZDVkLTlkNmEtMDFjM2Y5MTVkZWM5 lastKnownParent: CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configurati on,DC=cb,DC=cliffbells,DC=com isRecycled: TRUE whenChanged: 20160319092438.0Z uSNChanged: 4261 distinguishedName: CN=CBADC02\0ADEL:de85228c-...

-----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",

...succeded after transferring fsmo roles. 5) ran samba-tool domain demote -Uadministrator on FILER. 6) shut down samba on FILER, removed smb.conf, removed initscript 7) followed guidelines to cleanup any remaining references to FILER, it existed in AD Sites and Services, I removed it. I did not delete DNS references as FILER is critical in this network and must remain accessible. 8) rebooted FILER and CBADC01 Currently AD is allowing users to login to computers, all shares are dead because FILER isn't providing them and I can't set it up as a Domain Member to provide the shares again...

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le Jeudi 23 Janvier 2003 19:15, Rakotomandimby Mihamina a écrit : <p>HOWTO INSTALL icecast2 and ices - - Requirements * libxml2 libxml2-devel * libogg libogg-devel * libvorbis libvrobis-devel * libvorbisenc * libvorbisfile * libshout * icecast2 * ices2 <p>> On Thu, 2003-01-23 at 06:00, Michael Smith wrote: > >

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

...13) 8:15.065 H225 Caller:80f4688 H323 Clearing connection ip$localhost /25259 reason=EndedByConnectFail 8:15.065 H225 Caller:80f4688 H323 Call end reason for ip$localhost /25259 set to EndedByConnectFail 8:15.066 H225 Caller:80f4688 H225 Sending release complete PDU: ca llRef=25259 8:15.200 H225 Caller:80f4688 H323 Clearing connection ip$localhost /25259 reason=EndedByTransportFail 8:15.200 H323 Cleaner H323 Cleaning up connections 8:15.201 H323 Cleaner H323 Connection ip$localhost/25259 cl osing: con...