similar to: TDM400 stopped bridging outgoing FXO call

Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload, I'd hear lots of cracks in my echo, right? I'm not hearing that. Besides, a telephony card is not involved.

My SM56 (Motorola X100P clone) has echo as hight as 38%, according to fxotune -d. But when trying to take action, it fxotune simply says it can't. ./fxotune -i3 -vvvv Running with parameters: doset=0 docalibrate=1 dodump=0 startdev=1 stopdev=252 calibtype=2 waveformtype=-1 delaytosilence=0 silencegoodfor=18

How to compile smsq in 1.2? It is compile in 1.4 by default. It is included in 1.2.13, but not compiled. Any rule or method to make it? Yuan Liu

I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has

I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to

I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the topology: VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension | FXO ___ PSTN extension When I call a VoIP extension on that box (from a VoIP extension), voice is good. But when this box tries to bridge the call with a

I have an analogue line connected to an FXO module on a TDM400. I want to use the FXO line only for outgoing calls - incoming calls are answered by other equipment, but even if I don't specify an "s" extension, the FXO module still seizes the line for all incoming calls. Is it possible to configure the FXO module to ignore ring events completely? Wiki resources etc have been no

The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Yuan Liu

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

Does application Read() return a status? Console displays stuff, but show application read doesn't mention any status variable. Yuan Liu

Home someone can explain this: a Goto() command can walk within a macro, but if a digit is dialed from within a macro, the call flows back to the context that called the macro. Is there some way to "contain" the flow? Thanks. Yuan Liu

I mean how do I set failure condition in AGI? My script exits with code 0 upon success, and non-zero when problems occur - the standard *nix way. But Asterisk always report "AGI Script completed, returning 0", and AGISTATUS is always SUCCESS. Yuan Liu

Is there anything that allows a logical line to extend to the next physical line? Printed files are so hard to read with blind line wraps - and my printer doesn't even automatically wrap. Yuan Liu

This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with REN of 1.0B, a cheap speaker phone of 0.3B, and a cordless phone with marked REN of 0.0B. But it couldn't properly ring this 27935GE3-B (FCC ID G9H2-7930) cordless phone rated at merely 0.1B. Rarely, the phone will crack out an occasional weak and abrupt beap, but never a normal ring. Otherwise Asterisk and TDM400P

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called

Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel => 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with

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

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP----------------->PRI ------------> mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -------------- next