similar to: Blocking outgping caller id on a PRI E1

Is it feasible to enhance AMD to detect and report if the far end sends fax tones? I am guessing that, as it is using DSP to detect sounds and periods of silence, the same DSP can also report if a CED tone is sent. If it is feasible, is there a document describing the DSP interface that someone who is not familiar with DSP can use to get started? Neil Youngman Neil Youngman Developer Wirefast

On 11/01/2019 09:19, Administrator TOOTAI wrote: > Le 11/01/2019 à 10:12, Neil Youngman a écrit : >> A while back, I posted about detecting when a call was picked up by a >> fax machine. It was suggested that having a "fax" extension and >> "faxdetect=yes" would cause it to jump to the "fax" extension. This >> was not something I could

A while back, I posted about detecting when a call was picked up by a fax machine. It was suggested that having a "fax" extension and "faxdetect=yes" would cause it to jump to the "fax" extension. This was not something I could get to work. I have now created a very simple example. In sip.conf I have "faxdetect = yes". My example extension is: [test]

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

Package: xen-hypervisor-3.2-1-amd64 Version: 3.2.0-5 Severity: important The hypervisor loads OK but then fails to load my dom0 kernel. I get an erorr message "not an ELF binary" followed after 5s by a hypervisor reboot. /boot/grub/grub.cfg contains: menuentry "Debian GNU/Linux, Xen 3.2-1-amd64, Linux 2.6.22-3-vserver-amd64 (single-user mode)" { set root=(hd0,1)

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

Does SetCallerPres(Prohib) remove the ANI data from a SIP call or does it simply set a flag telling other devices not to display the data? In other words, could another system override that and see the caller ID anyway? The answer may affect how I handle 911 calls, so I'm very curious.

Is anyone else using the AMI with res_fax_spandsp.so for real-time status? I am working on migrating a FAX application from res_fax_digium.so to res_fax_spandsp.so. I have noticed that the spandsp module generates far fewer AMI status events than the Digium module and the generated events contain less information. For example when sending a fax there is no longer an event for every page. There

It seems that lustre puts its modules in /lib/modules/2.6.16.21-0.8-default despite the fact that my kernel is 2.6.16.21-0.8-smp ! uname -a Linux n32 2.6.16.21-0.8-smp #4 SMP Sun Sep 24 08:47:30 BST 2006 i686 i686 i386 GNU/Linux make[3]: Nothing to be done for `install-exec-am''. /bin/sh ../../mkinstalldirs /lib/modules/2.6.16.21-0.8-default/kernel/fs/lustre /usr/bin/install -c -m 644

Hi, Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on certain extensions. I have usecallingpres=yes in zapata.conf, and am using SetCallerPres(prohib) in my dialplan prior to the Dial command. No matter what I set SetCallerPres to the CID is still displayed. Is there something else I need to make this work? I can't just set the CallerIDNUM to null, as it is needed for

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]

Hi All, In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk. however, on the receiving phone, the callerid is still displayed. When i increment the logging of the pri with "pri set debug on span 1" on the CLI i also get the lower level debugging info from the pri.

Dear R and S users, I've been working on fitting finite mixture of negative exponential distributions using maximum likelihood based on the example given in MASS. So far I had much success in fitting two components. The problem started when I tried to extend the procedure to fit three components. More specifically, likelihood = sum( ln(c1*exp(-x/lambda1)/lambda1 + c2*exp(-x/lambda2)/lambda2

On Thu, Jun 18, 2015 at 1:26 PM, Matt Riddell <lists at venturevoip.com> wrote: > Did you buy the number from your carrier? Maybe it?s set on their side > for the trunk. > That's what I think too, but they are denying this. I think what's happening is they have a customer service guy interpreting logs (probably incorrectly). When I had a Century Link POTS line, I had a

When you set sendrpid=yes in sip.conf, a very nice feature is activated. When dialing an extension, the callerid of the dialed extension is returned back on the display of the calling phone. So if you call extension 100, you can see you are calling Ann (for example). I want to selectively disable the transmission of this information back to the caller. How can I do it? I tried setting

I have a little call recording script that I am running and it works fine, but I have one problem. I get CDR when a user calls into the extension, but I do not get CDR for the call that it makes outbound on # 17. Any idea why? Here is the extensions info: [default] exten => 2211,1,Answer exten => 2211,2,Wait(1) exten => 2211,3,Playback(/etc/asterisk/recording/getshop) exten =>

Are there instructions anywhere for setting up dom0 under Debian Lenny on amd64? Are there suitable Debian packages? James. _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

Hi; How can a user block his caller-id in Astersik? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/07ecf20f/attachment.htm

Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti