similar to: x100p/hangup detection issues?

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

All, If you currently own a Sipura SPA2000, avoid going to the sipura website and upgrading the firmware. I upgraded my SPA2k a couple of days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues with my SPA rebooting itself every 3-10 minutes for no apparent reason. I have been in touch with the *excellent* sipura support folks, and they are working with me to

Ok, I've googled for 15+ minutes, and have yet to find a usable answer, so I'm going to annoy everyone and ask here. I have, in my posession, a creative VoIP blaster. I have installed the fobbit LKM and I can see the device. Can I use it with asterisk in any meaningful way, shape, or form? I'd love to be able to buy an IP phone, ATA, or FXO card, but lack the funds at the moment

Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in

Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup("Zap/1-1", "") in new stack == Spawn

All, I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all* of my incoming calls are coming up as FAXes. I had to disable my fax extension because every call to my POTS line was getting redirected to my FAX machine. After removing the FAX extension, if I call my POTS line from my cell phone, I get the following: *CLI> -- Starting simple switch on 'Zap/1-1'

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

Hello Astrites, I was just about to send out a long email about not being able to detect hang-up with my CAC II FXO module on my PSTN POTS lines. I had tried many configurations. I had attached a toner to the line to see if I was getting this "disconnect supervision" signal after hang-up. (toner has a line powered light in the off position) It seemed that I was getting the signal,

Callerid issue 1) if you run ztmonitor on the fxo line & call in do you hear the fsk tone if yes then we beleive the CAC is passing fsk 2) in chan_zap->ss_thread around line 4154 (current cvs) if you get to the callerid_feed at least once then if you get to chan_zap->ss_thread->callerid_get around line 4163 (current cvs) does this parse fail

I use Sony (previously Sonic Foundry) Sound Forge, which allows me to save audio files in .w64 (Wave 64) format to get around the 2GB .wav file limitation. W64 was invented by Sonic Foundry, and is an open format as far as I know. The only programs I know about using the .w64 format at the moment are Sound Forge and Steinberg Nuendo, although there may be others out there. With increasing

*** Now you just need a "phone" or "station" (FXS) device and a method of connecting it. *** So... essentially I need to spend another $125 US on a Wildcard TDM400P so I can plug in an analog phone? On the X100P description it says: "The X100P supports FXS Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision). It can detect ringing and

Hopefully someone can help me. I've been running a windows 2000 Workstation and Server. This spring I added a Linux workstation (Red Hat 7.2 - Samba 2.2.3a) to the network. Took some work but that is running fine. smb.conf seems to be giving me what I want. LAN addresses are assigned with DHCP. I have had several people tell me that they have a network running with static addresses. I

Hello All, Just a question, I have an adit600 and I am looking for a way to pull the incoming cid into asterisk. Does anyone know if this is just not possible via t1? Or is it only available on PRI? Thanks, Greg

Hi, It seems like my ATA is making a ringing noise... (as it used to), but now (After the upgrade from 1.0.7 to 1.2) asterisk also is either making the ringing, or passing the PRI ringing from the telco on to me. Any suggestions on how to fix this?

I just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.

On 01/08/14 19:37, Reid Kleckner wrote: > What happens if you drop appending linkage? I think it will just work, > since you are already using a custom section, which will ensure that all > the data appears contiguously in memory. Thanks for the suggestion, but it still puts everything in a single .section statement. > Although, I do worry about what LLVM's alias analysis will

Hello, I am trying the setup the TDM01B card. 1 FXO. I connected it to a regular home line. in the /etc/zaptel.conf, I have fxsls=4 In the /etc/asterisk/zapata.conf I have: signaling=fxs_ls language=en group=1 context=default channel => 4 When I start asterisk, I get this error: ERROR[10376]: chan_zap.c:6584 mkintf; Signaling requested on channel 4 is FXO Loopstart but line is in FXS

Hi, I have two problems regarding the llvm.stackrestore intrinsic. I'm running on 3.0, but a quick test on trunk also showed the same behavior. First problem: --------------- I have code like: tmp1 = call llvm.stacksave() tmp2 = alloca [do some stuff with tmp2] call llvm.stackrestore(tmp1) [some other stuff] tmp3 = call llvm.stacksave() tmp4 = alloca [do some stuff

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