similar to: FW: Re: hangup detection

I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). After

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

I have a Centos 5.4 machine that has, for the past two weeks, apparently been shut off over the weekend. It's just sitting there turned off on Monday morning and when someone hits the power switch it comes right back on and everything works again. This happened last weekend, and again over this past weekend. Here is /var/log/messages from shortly before it apparently shut down this weekend.

Hello people, this is my first mail to this list. I'm new to asterisk and trying to set up an IVR. So far my dialplan works nice connecting with softphones, but I'm having problems to detect hangups on the analog line. Here are the details: -Clone pc with Ubuntu server 10.04 64 bits -Asterisk 1.6 -Openvox A400P, lspci -v says: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX

Hello, I have a problem with the Bristuffed version of Asterisk. I have tried Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have the same problem it seems: The setup: A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15. Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly. Grandstream gxp-2000 as a SIP phone, and a normal mobile

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

On Thu, Apr 25, 2019 at 10:03 PM Pankaj Gupta <pagupta at redhat.com> wrote: > > Dont support 'MAP_SYNC' with non-DAX files and DAX files > with asynchronous dax_device. Virtio pmem provides > asynchronous host page cache flush mechanism. We don't > support 'MAP_SYNC' with virtio pmem and xfs. > > Signed-off-by: Pankaj Gupta <pagupta at

On Thu, Apr 25, 2019 at 10:03 PM Pankaj Gupta <pagupta at redhat.com> wrote: > > Dont support 'MAP_SYNC' with non-DAX files and DAX files > with asynchronous dax_device. Virtio pmem provides > asynchronous host page cache flush mechanism. We don't > support 'MAP_SYNC' with virtio pmem and xfs. > > Signed-off-by: Pankaj Gupta <pagupta at

Hi all, This patchset makes some key modifications to the original ''stable page writes'' patchset. First, it provides users (devices and filesystems) of a backing_dev_info the ability to declare whether or not it is necessary to ensure that page contents cannot change during writeout, whereas the current code assumes that this is true. Second, it relaxes the

Hey everybody, I've had issues with detecting call disconnects on the Centrex system and an Adit 600. I've discovered that if I set the channel bank's FXO to lscpd, even though the CO says they don't support, that Asterisk can see the hangups. The catch? Asterisk can no longer seize the line. I've used this to my advantage, since I have 13 lines to work with. (5) +1

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

Users report that an extent map''s list is still linked when it''s actually going to be freed from cache. The story is that a) when we''re going to drop an extent map and may split this large one into smaller ems, and if this large one is flagged as EXTENT_FLAG_LOGGING which means that it''s on the list to be logged, then the smaller ems split from it will also be

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. The new versions of 'register' and 'asthostid' can be

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have been asked to ind out if there is a way to use asterisk to answere a voice fax modem so it can provide an answering service and record messages ? -- Gregory Machin gregory.machin@gmail.com www.linuxpro.co.za

Some people contribute to someone specific fs testing mostly, record some of them as Reviewer. Signed-off-by: Zorro Lang <zlang at kernel.org> --- If someone doesn't want to be in cc list of related fstests patch, please reply this email, I'll remove that reviewer line. Or if someone else (who contribute to fstests very much) would like to a specific reviewer, nominate yourself to

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin