search for: skycomuk

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

Hi, The subject says it all. Is it possible to put the IP address of the peer in the CDR records? Using AGI maybe? -- Kind regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbressers at signet.nl

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

...:p></o:p></span></font></p> <div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span = style=3D'font-size: 12.0pt'>On Mon, Oct 18, 2010 at 4:39 PM, Gareth Blades &lt;<a href=3D"mailto:list-asterisk at skycomuk.com">list-asterisk at skycomuk.com</a>= &gt; wrote:<o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span = style=3D'font-size: 12.0pt'>Use camailio or opensips as the regi...

...cles/damocles_mini_en.html). of > course, the damocles will have to drive a high-power relay. > > the damocles can be driven via snmp, so you'll have to simply call the > snmpset unix standard utility > > On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades > <list-asterisk at skycomuk.com> wrote: > >> Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only >> control 750W so you will probably need to get it to control a more >> powerfull relay as a heater is going to take a lot of current. >> It can be controlled by a virtual serial...

Hey, all. I'm trying to do some fun with auto-provisioning of Polycom phones, and one thing that would make life easier for me would be if I could have a per-phone sip.conf file. If not, no biggie -- but if there's a way to do an include (as per extensions.conf) or something, that would be great. I've gone through docs, and an older version of "Asterisk: the Future of

Greetings fellow listers, I have an application where I have approximately 300 files that I playback individually or in blocks to simulate "text-to-speech" in a "less mechanical" voice than normal Allison files provide. These files are presently in GSM format and sound pretty good when I play them on my computer speakers or on my in-house

...file it addes in an > iptables rules to block the traffic for a period. > > On debian you can apt-get install fail2ban and on centos/redhat yum -i > fail2ban > > Thanks > > Kenny > > ----- Original Message ----- > From: "Gareth Blades" <list-asterisk at skycomuk.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users at lists.digium.com> > Sent: Tuesday, 29 June, 2010 4:12:42 PM > Subject: Re: [asterisk-users] Find a way to block brute force attacks. > > Rodrigo Lang wrote: > > Hell...

Hello, I want to create channel bank in this case: "channel bank" |-----------------------------------------| | FXS,FXO<----->TDMoE<--|---------------------------------->Asterisk |-----------------------------------------| How can it?

Hello, Is it possible to install Asterisk on Vmware(centos) from source. Is there any difference or disadvantage for this compared to asterisk running on physical machine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100811/05a14968/attachment.htm

I am running asterisk 1.6.2.6 and have configured hints for our extensions and have a couple of Aastra 6755i test phones. The phones register fine but 'core show hints' shows the lines as idle even if they are in use. I read the wiki and see mention about needing to set call-limit in asterisk 1.4 but that has been depreciated in 1.6 so what is the way it should be done in 1.6?

We are a telco so when we receive calls via ISDN and the number is witheld we see the correct presentation value but also still see the actual callers number in the callerid(num) variable. I am trying to forward some of these calls over to one of our other boxes via SIP but I have found that if the number is withend then the sip packet contains :- From: "Anonymous" <sip:Anonymous

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

I have upgraded Asterisk from 1.4.22 to 1.4.30 and I have noticed I am getting a lot of errors like this on the console :- ERROR[23912]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe I have tracked it down to a perl AGI script which performs our own CDR recording. It is called before the start of the call, once answered and again when the call is hungup. It works fine when