similar to: Buffering DTMF input

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

Hi, I am lost trying to figure out how to enable overlap dialing for calls coming in and coing out on a pri span. DISA looked promising at first, but does not seem to support overlap dialing. Just picking up a call by and trying to dial out does not seem the way to do it either. I tried: [dialincontext] exten => 12341234,1,Goto(dialoutcontext,s,1) [dialoutcontext] exten => s,1,Wait,1

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

Did anybody figure out how to make dial detect a busy on a zaptel channel on a pri interface when using overlap dialing? According to the documentation dial should return to priority n+101, if the called party is found to be busy. I can see a DISCONNECT message with "user busy" coming from the network when I turn on pri debugging, but the dial application does not seem to notice.

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

Hello, I try to write a bash skript and I want to use the variables from my bash skript into R. Ist that possible? My bash skript creates lots of *.data files. I want forward these files directly into R (in x.data.bz2), so that R creates a few data automatically also in PDF. For example: bash created files: hello.data , world.data how R created these files in pdf? please look my plot.R

When I was trying to run mutiple E1s over TDMoE, the zaptel would drivers complain about too little memory, whenever I added more than 31 channels. Requesting 62 channels in a dynamic span gave me ... span creation failed: Cannot allocate memory upon loading the zaptel drivers. How would you go about running, 8 or 16 say, E1s over TDMoE? Would you create multiple dynamic spans or just one large

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

Hi everyone, any chance * could be used as a b2bua without forcing the media stream through the same box? I would love to do some computing on incoming calls, do things like setting another callerid and the forward the call to another sip UA - all without any audio traversing the * box. Any ideas? Thanks, Thilo

[ZapRAS triggering a kernel panic] >> Kernel panic: Aiee, killing interrupt handler! >> In interrupt handler - not syncing >> HDLC Receiver overrun on Channel Tor2/0/2/25 (master: Tor2/0/2/25) > > Hrm, I haven't seen this before. Please contact me off-list and I'll > give you more debugging instructions that may be helpful, as well as > enquire additionally

Hi, I have problems to save printer settings like changing the size from letter to A4. This happens for user settings and default printer settings. The settings are always resetted to default values like "letter". The system uses samba 3.4.11, I also did some tests with samba 3.5.8, but nothing changed. After downgrading to 3.3.2 everything worked as expected. I didn't change any

Hello, Does anyone know what driver to use with an "Eicon Diva Server 4BRI" card? My modem.conf: -------------- [interfaces] context=remote driver=i4l msn=16453 device => /dev/ttyds01 -------------- I get the following error: -------------------------- == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem Driver) WARNING[8192]: File chan_modem.c, Line 383

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

Hi all, Is there any way to get a skript or some commands executed on connection(!) of a host (OS is Linux)? I'd like to set a host route on my router (my LAN server running Debian Linux) when my notebook (Debian Linux, too) connects to it. As I understood tinc-up is executed on startup of the tinc-daemon which doesn't fit for me. The hostroute is ONLY to be set if the Notebook is

For all who are interested: A quick review of CeBIT 2005. :-) CeBIT was a very successfull event. Most of the time, the asterisk-booth was crowded with more people than we could talk to. We had with us a demo-installation including different IP-phones, digital and analog phones as well as a Siemens HiPATH PBX to which our Asterisk-server served as a VoIP-gateway, and many people were impressed

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different

Hi All, I hope someone can give me a hint. We try to reload the asterisk dialplan config using ansible command module. Using this we just trigger asterisk -rx "dialplan reload" Now we want ansibe to fail if there is something wrong in the dialplan. If we put a bad config in extensions.conf dialplan reload prints some warning in Asterisk console but the command "asterisk