similar to: PCI Master Abort

Could someone please assist me soonest. when i run asterisk with one X101P card and a S400P card asterisk will run but the card will not work.this is the message i get: [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_mysql.so] =>

Hi, Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk? I'm especially interested if you've used it with a TE405P or TE410P. Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

hello all, I still experience the random off-hook on my fxo cxhannels, i am using a zplex-10b channel bank. which does not allow me to call out. The situation still persists...this is what i have in the zapata.conf [channels] context=internal context=incoming context=default usecallerid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

hello * users, i am using a zplex 10b(8fxosand 16fxs) connected to a T100 digium card, RH8.0. soon after running * i get this warning "file chan_zap.c,line 4155 (ss_thread):CallewrID returned with error on channel 'zap/7-1' then executes the other applications itself. --executing Wait("Zap/7-1","1") in new stack. -- executing..... it gives the same warning

can somebody that has these hardwares(X101P and S400P) working on his asterisk system please assist.................. you can send the solution to austino@skannet.com..... the error message is what i have below. ---------- Forwarded message ---------- Date: Thu, 1 May 2003 21:01:21 +0100 (WAT) From: austino@skannet.com To: asterisk-user@lists.digium.com Subject: Error working with X101P and

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

i am using RH8.0,when i try to load the zaptel driver using "modprobe zaptel" it give an error "cant locate module zaptel" also when i do "modprobe wcfxo" it gives "can't locate module wcfxo" when i try compilling zaptel i get "make ***[zaptel.0] Error 1 what am i doing wrong someone please help! -- Augustine GDES 18b oshin road,kongi

Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I?ve been playing around with ?sip show channelstats? but can?t other than measuring the packet loss I don?t really know what I?m supposed to be looking for

Ok. I just got a TDM10B and it is in with my X100P. So as it says in the provided instructions, I used the command modprobe tor2 I get an error message saying that there is no such device. My zaptel.conf looks like this: fxsks=1 fxoks=2 So I load the X100P first. (modprobe wcfxo) Then I load the TDM10B (modprobe tor2) Then I'm told that the device doesn't exist. Please help

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a

Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before?

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

The problem is the following # make docs Make: Don't know how to make ../src/library/*/man/*.Rd. Stop. *** Exit 1 (ignored) # I am using a DEC alpha 200/4/233 with Digital Unix 3.2D My previous version of R was R-0.50-a1. I have installed around 3 or 4 previous versions on the same system and never had a problem (but, then the Makefile was very different). The first make to compile

The problem is the following # make docs Make: Don't know how to make ../src/library/*/man/*.Rd. Stop. *** Exit 1 (ignored) # I am using a DEC alpha 200/4/233 with Digital Unix 3.2D My previous version of R was R-0.50-a1. I have installed around 3 or 4 previous versions on the same system and never had a problem (but, then the Makefile was very different). The first make to compile

Douglas Bates wrote: >From 0.62 onward you should not have to create a symbolic link in >/usr/local/bin. It should be that you can run > cd $RSOURCE > ./configure --prefix=/usr/local > make install >and you will end up with the R script installed in /usr/local/bin and >all the files needed to run R in /usr/local/lib/R. > >Can you tell what the prefix is set to after

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing