similar to: DTMF tones too long, for once

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for "Always" works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to "No Answer" means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a

We're running Asterisk 1.8 on a 32-bit Debian machine, and it has been fine for some time now. But! We've got such a incoming call volume over the few weeks that we'll have Asterisk occasionally restart itself. My hunch is that it is in part memory pressure. I can't add RAM and have it help, because it's 32-bit. I intend to move to a 64-bit machine, but I was hoping to

Greetings All! The Asterisk Call Manager works great. But I have one question for anyone who has used it. I cannot get the system to send some DTMF tones down the channel once the call has been made. Below is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system

Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to run a small home PBX with two analog telephones connected to a Linksys ATA using SIP. It works great (except for some Bluetooth adapter bugs that I am still trying to beat...seems the misaligned audio detection still needs work), but I have encountered an interesting issue. If I am using an automated system that accepts input

Just to add.... P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0 P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] PH_CONTROL: channel:1 oad2:07nnnnnnnnn dad0:820055 P[ 1] --> DTMF

If you're an asterisk 1.6 user, and use the 'Directory application', have you noticed that the first keypress is always missed if you press it during the part of the announement where alison says "using you touch tone keypad" If this includes you, have a look at mantis bug https://issues.asterisk.org/view.php?id=16409 If you are keen, please apply the patch and report

For the last few months we have intermittently been experiencing some very strange registration problems with certain polycom phones. Here is some background information: I have about 150 Polycom Soundpoint IP 600s, 601s, and 650s spread between 8 servers at different locations. Each phone is on the same network (and subnet) as the server it connects to. There is no NAT or anything else strange

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. Locally, if I call from an

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

When my Asterisk box is called from my company's standard type of phone, it does not detect normal button presses on the keypad. You can use the phone's 'touch tones' feature to send data to the box, but that is a bit cumbersome. My Norstar voicemail system (that is being replaced) does not have this limitation. Is there anything I can do on the Asterisk side? Thanks, Bob

I'm having trouble getting the quota plugin to work with deliver. The quota in the database is not getting set. I am using dovecot-1.0-rc15. This server is not yet in production. Some values below are because I'm trying to test with the simplest possible case, and some because it's reading from a different database. Also, I noticed that "dovecot -n" doesn't display any

configure.ac | 2 doc/Makefile.am | 3 libswfdec/swfdec_as_object.c | 17 ++++ libswfdec/swfdec_as_object.h | 5 + libswfdec/swfdec_keys.h | 1 libswfdec/swfdec_pattern.c | 5 + libswfdec/swfdec_player.c | 100

Hi, Below is my case. phoneA (PSTN) phoneB (SIP) phoneC (PSTN) phoneA --> asterisk --> phoneB phoneA (music on hold), phoneB --attended call transfer--> phoneC phoneA --connect-- phoneC after phone B hangup phoneA type some keys in keypad but phoneC always has wrong dtmf detection. In my case, I would like to know any factor that will cause the wrong dtmf detection.

Hi all, I managed to get most functions of Asterisk operational with Messenger. Accessing menu's have I done with DTMF generated tones by Windows Media Player, because this seems to be the only program without sound-card conflicts with Messenger. The tones are mixed with microphone-input. Does any of you know a nice dialer with a graphical keypad that does not grab the sound-card exclusively.

HI, I am trying to write an AGI that will authenticate users befoer the call is allowed to proceed. Befor you ask, i have tried using the authenticate() function but it does not work for me as this function messes up call accounting (authenticat() awayas awnseres the channel, thus causes CDR to bill for 'ring' time). the AGI will (1) playback a voice prompt over an unawnsered channel (2)

I wrote a small program to monitor keyboards found on /dev/input/event* It works great but I always get EAGAIN from my read() function. google says this is normal when open() is used and O_NONBLOCK mode. I know this is slightly offtopic but I was wondering if the centos gurus that also program know anything about this. My goal was to not be eating CPU cycles with a small program that just