similar to: New Member, saying Hi. :)

Has anyone on here used kiax before? I am asking because I have it installed on several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones. The problem I am having is this: when I use kiax to call someone else, they get some kind of background music playing while I am talking to them. I have called from kiax to

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

I have asterisk running more or less ok but I would like to turn off call waiting and be selective about the incoming sip connections. This is running asterisk 1.2.8 with a fxs and fxo card and a configured voip (sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk. Problem 1) if someone is on the phone already and another call comes in for an already engaged extension I

I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten => s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

I have been looking around some and I can't seem to find anything which will answer my question. If I have two Asterisk boxes in different locations which are linked to each other over the internet, can I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line

Hi all, Can anybody tell me how we can create RSS feed using Ruby on Rails. How do I go about constructing this RSS file? Can I find an example in any site? regards, Prasad -- Posted via http://www.ruby-forum.com/.

Howdy folks, I''m trying to boot Xen 3.0 on a Centos 4(.2) system with root on a software raid1 volume. When booting the Xen kernel, ''raidautorun'' in the /init nash script fails to assemble either the root md or the raid5 md containing my LVM volumes. If I move root to /dev/hda1, the system boots fine. raidautorun still fails to assemble the md devices, but mdadm

I have 2 identical AudioCodes MP-112s. They have the same config except for the SIP usernames/passwords and the device IP. The configs in extension.conf and sip.conf are also identical. On one box, when I pick up the phone, I get a fast busy and the logs/debug show an automatic hangup. On the other device, I can make calls without a problem. I can even call the phone that can't make a

On 10 Jun 2020, at 23:18, @lbutlr <kremels at kreme.com> wrote: > IF it?s not permissions you need to provide doveconf -n output. Bloglines for any fall, panic, or error level events at a minimum. Apologies, I did not see the attachments. Will look on a real screen later.

We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of

Hi folks, I took a break from Mongrel today and instead whipped up an Atom feed generator for my site. People who love or utterly hate my essays will lover or utterly hate to subscribe to it. http://www.zedshaw.com/feed.atom Let me know if it isn''t working well in your favorite feed reader. -- Zed A. Shaw, MUDCRAP-CE Master Black Belt Sifu http://www.zedshaw.com/

I don't see a method to detect the success or failure for the Record CMD. I'd like to know the reason why the recording ended Am I wrong? exten => recordmsg,1,Noop() exten => recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180) Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, since one or two month I have a problem with Dovecot not updating the read-flag on mails using IMAP. I receive new mails, but when reading, they still unread on other devices (but shown as read on the first device). When I remember correct, this problem doesn't come with an update of dovecot, but just occured at some time. But I've updated dovecot since then a few times.

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Hi, I'm looking for a great tech support person to take over the admin of our asterisk system. If you are a networking person as well, with some experience in firewalls and desktop support even better. The system is a multi-group system with IVR, Follow-me dialing, voicemail, and conferencing. Multiple SIP providers are in use. If you feel you can help us, or can recommend someone that

Hi, What USB headset would you recomend? We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Thanks! Regards, Jan

I want to set up a test system at my house using a FXO card (one of the X100P cards) and a FXS device. I am going to have the one line running into the machine and the house phones running off of the FXS port. I will be installing a softphone on my laptop and will be using Hamachi to tunnel back to my Asterisk machine while I am out and about. This way, I can leave my firewall intact and still

>Easy to do on the Linksys PAP2, if that helps. The functionality > probably depends on the make and model of the phone... maybe if you gave > those details as well? > > James > Well, there's the rub. I don't have any of the hardware yet. I am looking at the various options before buying anything. I know that sounds like an odd way to do things: research it before