similar to: hangup detection

Hello - My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesn't disconnect right away. Any ideas on how to resolve this? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I have this little problem here and i really don't know how to solve it. This is the scenario: I've setup a IVR, using my mobile phone I call my asterisk server and after pressing "1" the call is directed to my softphone at extension 100. The phone at extention 100 will ring until a certain time, and my mobile phone will cut off due to no one picking up my call. However,

I'm filtering that is being deliverd to postfix mail server with amavisd-new . I want spam with spam f level 1 - 8 to ad a tag any everything above to be delete is this posebol? If yes how? Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax: +3130-6342433 Prive Tel: +3130-6372769 Mob:

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? Best regards, Han -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f1043e19/attachment.htm

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk

Bruce et al. I'm posting a new thread with the "Top Posting" subject so I won't draw complaints about "hijacking" the 4-port thread. Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If I'm actively following a thread, the most-recent information appears at the top

Has anyone seen good scripts or documentation on Voicemail groups? We are looking to have a system where you can send a voicemail to multiple mailboxes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/777a7b83/attachment.htm

asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org ________________________________ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph.

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Hi, I tried to install R-2.4.1 on cygwin system. "./configure" succeeded, but make failed. Below, I provide the output from the process: error message, and info from configure output, in that order. I appreciate that someone can guide me (technically in-sophisticated) through this process. Again, thanks for your help. Michael Niu (1). Output from make make[3]:

I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ?f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567---------------------------- registered to Asterisk

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing

My site : https://radiobiscottes.studioo.fr/ -------- Norbert Deleutre  <http://www.lmgc.univ-montp2.fr/perso/norbert-deleutre/> P 0467149655 UMR CNRS 5508  <http://www.lmgc.univ-montp2.fr/> A Campus Saint-Priest/Montpellier > Le 25 nov. 2020 à 13:04, Damien GENESTE <d.geneste at illud.fr> a écrit : > > Hello Norbert, > > I don't know how i can help

Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Dear Xen community, I have Xen 3.0 installed on RedHat Linux Enterprise RHEL4U2. "xend install" runs fine with no error messages. However, when I start "xm cr guest-vmx.conf" I do not get any new window open for the new guest OS. "xm list" shows that the vmx has started and seems to be working fine (just for testing, when I type "xterm" an X window

Hi I know this gets covered fairly regularly, but I've had a search through the archives and can't find anything dealing with this specifically - apologies if I've missed it though. I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the