similar to: Zap & ANSWER the Call

Hi list, I''m playing with tc and found a strange behavior when I try to delete filters. For example, this simple scenario: tc qdisc add dev eth1 root handle 1: htb default 100 tc class add dev eth1 parent 1: classid 1:1 htb rate 128Kbit tc class add dev eth1 parent 1: classid 1:2 htb rate 258Kbit tc class add dev eth1 parent 1: classid 1:100 htb rate 32Kbit tc filter add dev eth1 parent

Hi Folks, I've been having troubles with a DSL router (DSLink 200E) and SIP phones. When I put any SIP phone (software or hardware) to work behind that DSL router, it completely freeze. I ready tech specs of that DSL router and it says that SIP protocol is supported. ie. I tested two DSLink 200E with the same results. Does anyone has any idea? Thanks in advance. --

Hi Folks, I've been searching for an specific feature on asterisk and I found an e-mail from "John Todd" asking for the same thing. http://lists.digium.com/pipermail/asterisk-users/2004-May/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a

> > > > I would be grateful if anybody could tell me what I should > > tell Verizon > > in NJ so they would enable "disconnect supervision" for my lines. > > > > Apparently "remote hangup notification" or "disconnect > supervision" or > > "calling party control" is NOT the magic phrase for them. Although >

Hi list. I have Asterisk installed on a Debian 1.8 6 64-bit. What happens is the following, some channels are not being hangup properly. They run the hangup in dialplan, but the output of the command "core show channels" shows several channels with status "rsrvd." Checking the server's memory, the "top" command shows multiple processes and stopped using the

Hi All, Which is the best signalling to use when connecting an FXS inteface on a TDM400 to a standard telephone. I see that all examples use fxo_ks, but it is my understanding that kewl start is really designed for connections to the CO so that hangup etc. can be detected. So does it make any sense to configure a telephone for fxo_ks? Or should it be configured for fxo_ls? Regards Garry Taylor

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

I have the following setup.. [extensions] ; all extensions defined here. exten => 1234,.... exten => 1235,.... [dial-out] ; PSTN dialout config ignorepat = 9 exten => _9,.... exten => h,.... [local] ; phone context in sip.conf is here.. include => extensions include => dialout The question is where will the "h" extension be active?? it appears to run for ALL,

Hi, I have two small companies that are going to be sharing a * box. I have 2 TDM400's with 4 fxo ports each. Each company has its own sales person and they would like the sales people to always show their own caller id and have their own lines ring directly to their phones. Company 1 sales person uses the 1 port on the tdm400 and company 2 sales person uses the 2nd port of the tdm400.

Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good).

Good evening everyone, I went to change the Rails version of my project, and saw that many things have stopped working. As the "rake routes", which is giving the following error: rrmartins rodrigo @: ~ / Documents / vota_prato $ rake routes rake aborted! no such file to load - tasks / rails (See full trace by running task with - trace) What do you think you can be? thanks -- *

My specs work fine with this in my Gemfile.lock: grep rspec Gemfile.lock rspec (2.11.0) rspec-core (~> 2.11.0) rspec-expectations (~> 2.11.0) rspec-mocks (~> 2.11.0) rspec-core (2.11.0) rspec-expectations (2.11.1) rspec-mocks (2.11.1) rspec-rails (2.11.0) rspec (~> 2.11.0) rspec-rails (~> 2.11) After "bundle update

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

Is there any way to prevent Rails 3 from using an ORM? In Rails 2, it used to have a description in environments.rb explaining how to do that. In Rails 3, is there any way to tell it to not use any database? I was trying to make some benchmarks from situations that don''t require a database... Thanks in advance, Rodrigo. -- You received this message because you are subscribed to

Hi all, I need to calculate a multidimensional integration on R. I am using the command "adapt" (from library adapt), although sometimes I get the following error message: Ifail=2, lenwrk was too small. -- fix adapt() ! Check the returned relerr! in: adapt(3, linf, lsup, functn = Integrando1) I guess it happens because the domain of integration is too small, although I tried a

Hi i have a big problems with my asterisk .. i use a Digium TDM400P for connect a analog line. And not all time (i don't know why) he don't see the end of the call and anyone can call me (occuped) For that's work, i am disconnect the phone cable and it's good anyone have a idea ? bye