similar to: fsk callerid with DTAS start?

After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have "phone company" compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

Hi, I'm currently having problems detecting FSK BT (UK) caller id in our API (Pika boards). I have a recording to test on but it is giving me checksum errors. I'm wondering if someone from UK using BT lines could send me a recording with the FSK signal so I can have more data to work on? If you have something, please send it to paulo.astuser at gmail.com. I appreciate all help about

Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

On Mon, 23 Jul 2018 14:48:00 +0800 d tbsky <tbskyd at gmail.com> wrote: > 2018-07-22 17:44 GMT+08:00 d tbsky <tbskyd at gmail.com>: > > 2018-07-19 23:59 GMT+08:00 Rowland Penny via samba > > <samba at lists.samba.org>: > >> > >> Please see inline comments. > >> > >> On Thu, 19 Jul 2018 23:44:48 +0800 > >> d tbsky

On Mon, 23 Jul 2018 18:34:36 +0800 d tbsky <tbskyd at gmail.com> wrote: > 2018-07-23 18:01 GMT+08:00 Rowland Penny via samba > <samba at lists.samba.org>: > > On Mon, 23 Jul 2018 17:19:07 +0800 > > d tbsky <tbskyd at gmail.com> wrote: > > If, as it sounds, you were altering the users primaryGroupID > > attribute, then you should not have been doing

All I would be interested if the current cvs zaptel, has fixed any callerid check sum fails on x100p's. Mark worked on an issue i had with a false 2nd ring detection from the fxo ports on a CAC Adit 600 interfaced to a T400. He added logic to the zaptel driver to ignore a 2nd ring detection if it comes to close in ms to the first ring detection. A false 2nd ring detection could cause the fsk

2018-07-23 16:04 GMT+08:00 Rowland Penny via samba <samba at lists.samba.org>: > On Mon, 23 Jul 2018 14:48:00 +0800 > d tbsky <tbskyd at gmail.com> wrote: > >> 2018-07-22 17:44 GMT+08:00 d tbsky <tbskyd at gmail.com>: >> > 2018-07-19 23:59 GMT+08:00 Rowland Penny via samba >> > <samba at lists.samba.org>: >> >> >> >>

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, 23 Jul 2018 17:19:07 +0800 d tbsky <tbskyd at gmail.com> wrote: > 2018-07-23 17:02 GMT+08:00 Rowland Penny via samba > <samba at lists.samba.org>: > > On Mon, 23 Jul 2018 16:46:50 +0800 > > d tbsky <tbskyd at gmail.com> wrote: > > > >> 2018-07-23 16:04 GMT+08:00 Rowland Penny via samba > >> <samba at lists.samba.org>: >

2018-07-22 17:44 GMT+08:00 d tbsky <tbskyd at gmail.com>: > 2018-07-19 23:59 GMT+08:00 Rowland Penny via samba <samba at lists.samba.org>: >> >> Please see inline comments. >> >> On Thu, 19 Jul 2018 23:44:48 +0800 >> d tbsky <tbskyd at gmail.com> wrote: >> >>> thanks a lot for the quick help. I remember in old days it happened

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

2018-07-19 23:59 GMT+08:00 Rowland Penny via samba <samba at lists.samba.org>: > > Please see inline comments. > > On Thu, 19 Jul 2018 23:44:48 +0800 > d tbsky <tbskyd at gmail.com> wrote: > >> thanks a lot for the quick help. I remember in old days it happened >> sometimes. but after upgrade rhel 7.5 (from samba 4.6.x to 4.7.1) and >> samba DC

Hi: I found I can setup password for uspmon. but upsc can connect to any upsd without authentication. although the ups data is not very confidential, but I would like not to expose it to anyone who can connect to server. is there any method to harden upsd? thanks for hint. Regards, tbskyd

hi: our current connection is below: sip phone<--->asterisk<---->alcatel PBX<---->PSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3.

HI: i just trying memdisk that comes with syslinux 3.35. i try to load a dos image, which contains ghost verison 8.3 and ghost version 11. both version of ghost hangs when start-up. memdisk that comes with 3.40 pre is ok. and many earlier versions is ok too. i try some parameter like "safeint","raw". but none work. i use memdisk with pxelinux, below is part of my

hi: when testing nut in our environment, we found something that nut maybe tune for "stale/dead ups" situation. currently the "dead ups" are assume alive(eg: host shutdown unnecessary), unless it is in the "OB" state before going to stale. our environment (ServerA + ServerB forms a cluster): ServerA-> usb to UPSA -> two PS power by UPSA and UPSB