similar to: Queues and RingInUse

Just a silly question that I'm not sure. Ringinuse is working with IAX in 1.6??? like in sip?? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090306/fa56ebfc/attachment.htm

Dear all i'm planning an upgrade of some asterisk installation from 1.4.32 to 1.6.0.28 (as i think it should be the most stable now). Reading the UPGRADE-1.6.txt file i've noticed that: * SIP: The "call-limit" option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the

I'm setting up a queue for two independent operator phones that are capable of answering multiple calls at once. It's currently working with the following settings and Asterisk 1.4: queues.conf: [telefonistas] strategy=roundrobin ;strategy=leastrecent music=default timeout=60 retry=0 maxlen=0 wrapuptime=0 ringinuse=yes autofill=yes joinempty=yes member => SIP/8899 member =>

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

Humm... Seems like my message didn't make it. Here we go again.. /Johan -------- Original Message -------- Subject: Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer <lists at jttech.se> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> On 2011-06-05 19:54, virendra

Hi, I've a working samba configuration with shares mapped to groups with "valid users = @smbusers". It all works fine with tdbsam as a backend. Now I've switched to ldapsam as a backend. I've a working openldap server that my windows machines authenticate to. Each user is also a linux-user but linux does not use ldap but /etc/passwd. However the groups in does not work and I

Thank you all for the comments and suggestions! I got the link from Garbor to work, but that is the old r-travis approach (using C). I tried the same approach with native R Travis build but unfortunately it did not work. So I contacted Jim Hester and he told me that they are now actively working with implementing multiple R versions in the native R builds. /M?ns 2016-01-18 15:17 GMT+01:00 Dirk

Quoting Tim Litwiller > to connect to the speaker system I either need to trigger a ring on a > analog line to the phone interface on our speaker system, it picks up > on > the first ring, or we can manully push a button that picks up the > line. > If we do the second we would have to have something in asterisk > connect > it to the conference when it picks up. We just

Hi All we are using Asterisk 1.6.0.9 version.try to use Minivm for voicemail, but having following problem. 1.How any extension let's say 7001 can access his voicemail box from his deskphone, any config or dial plan example is there. What kind of config require in extension.conf,minivm.conf & sip.conf 2. same way for MWI on respective IP Phone what we require to config in

Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to

Shortly, I'll be purchasing a Sangoma A102. I'm wondering if I should spring for the hardware echo cancellation circuit or not. Upon initial implementation, the 2 T1 Ports will be used as a passthrough as we slowly transition off of a legacy PBX. Eventually, we'll only be using one of the ports, and will be providing VoIP service to a bunch of SIP deskphones. So - with that usage

Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the

Hi All, I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is

I've been asked for recommendations for a small call centre, an ethernet SIP deskphone with a wireless headset. Similar approach would be a mobile phone with bluetooth head set. Either I've not looked hard enough, or there isn't much on offer. Alec Davis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history

I have (more than 1) provider that I receive calls from using IAX, and I have 2 IAX deskphones, all work fine except for some reason with 1 provider, when the call comes in, it doesn't match up with the incomingcall context. (A bit worrying, since I don't want people to be able to relay calls off me.) in iax.conf I have: [ipcomms] type=user nat=yes dtmfmode=rfc2833 host=71.16.179.149

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

For the 3rd consecutive term, the US Senate has introduced the "Truth in caller ID Act of 2009". It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out