similar to: GUI: Not Found. Move along

Hi people! I am not getting really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html

i have the problem in the hardware linksys pap2t, I am install asterisk with asterisk-gui and work fine but the hangup the phone (linksys pap2 t), no tone and sound like tu,tu, tu , tu , tu , tu , tuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuu what is the problem with phone ??? add param special??? Note: i am mark number phone and wait ... sesonds and call. thank you. -------------- next

Hello Asterisc-Users List, I am new to the list. I joined with a question in mind: How would you set up an asterisc box so that: (A) Someone dials a number (B) They are presented with a menu (C) Entering a number, like 1, connects a call to me. (D) I am on a mixing board, running an internet radio show. I want to run asterisc into the board, and run an output from the board to asterisc. Is that

Hi, List, I am looking for a cheapest (and therefore most funny) way to attach GSM card to my asterisk home box. Needed features: Calls+SMS in/out one or two SIM cards (ports) Should I try looking for a GSM PCI card that is compatible with linux/asterisk, or GSM USB card, or modern full-blown SIP GSM gateway (with ethernet)? Maybe an ordinary cell phone with USB interface and mangling with

hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install >> its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html >> now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is

On our CentOS 7 servers I see there is a new kernel available when doing yum update: kernel 3.10.0-1062.12.1.el7 kernel-devel 3.10.0-1062.12.1.el7 However I am not able to find any announcement at redhat.com for this kernel. The only announcement I find is this from CentOS: https://www.mail-archive.com/centos-announce at centos.org/msg11573.html However that CentOS announcement links to

Well... THANKS to wine developers!! Never thought i could play WOW on salckware! Actually it's running a LOT smoother on wine as it did on xp. However the only problemn is that the cursor can't reach the far bottom on the screen where my bar is. So i can't put my spells in the quick slots. Any suggestions about it? Is there someway to solve this minor issue?

Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports?

Hi, I've been configuring AsteriskNOW from the GUI but could it be that the GUI isn't working properly? because when I make a queue and add a few agents, and when I call the queue none of the phones ring. The queue is also configured at "Ringall" I checked the queues.conf file and the settings matched. I also noticed that the agents I made in the GUI, that they were not written

Hi, I''m using like this var pars = ''table=''+this.config[''table'']+''&op=update&id=''+id; new Ajax.Request( this.config[''url''], { method: ''get'', parameters: pars,

What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre: http://www.asteriskexperts.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear list, I want to update my prabclus package which I haven't done for quite a while. In the previous version, I had .dat files in my data subdirectory, which I read using .R files. Now R CMD check gives me a warning that .dat files are no longer accepted there. So I changed my filenames to .txt, but actually some of these files are only there in order to be read by .R, not in order

Hi , I still can't figure this out. We have two different network connections with a asterisk servers each. I want to make calls between asterisk servers. (i.e.) i want to call from a number in Asterisk server A to a number in Asterisk server B. I want to implement this using SIP. Will be very useful for me if someone helped to make this work. Thanks in Advance. Regards

Hi everyone, Is it possible to call from your Asterisk server to the Skype network? i.e., let's say I would like to call from an extension from my Asterisk PBX machine to a Skype account, is this possible? I did a little bit of searching and they were talking about that's only possible with windows machines, is this true? greetz, Tim

Hello, On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a way to enable HTTPS. Asterisk is running as asterisk:asterisk: asterisk 11097 0.3 6.7 741352 67984 ? Ssl 17:53 0:06 /usr/sbin/asterisk -g -f -p -U asterisk # cat /etc/asterisk/http.conf [general] servername=Asterisk enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089

Hello, Le lun. 6 janv. 2020 à 19:01, Olivier <oza.4h07 at gmail.com> a écrit : > May I add I could successfully (if pjsip show transports has any meaning) > add a PJSIP TLS-transport with: > > [transport-tls] > type=transport > protocol=tls > bind=0.0.0.0:5061 > cert_file=/etc/asterisk/keys/asterisk.crt > priv_key_file=/etc/asterisk/keys/asterisk.key >

I am trying to connect Asterisk and Openfire together, but it's not yet working completely... I don't know for sure if my manager.conf-file is set correctly. I use this manager.conf file just to let Openfire talk to Asterisk... [general] displaysystemname = yes enabled = yes webenabled = yes (is this necessary for Openfire ???) port = 5038 bindaddr = 0.0.0.0 [openfire] secret=XXXXXX

Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over

I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname => mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called "sysname" in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The