similar to: asterisk take 99% of CPU resources

Hi, I've connected a TE110P from digium with a 2 E1 to a Siemens PBX using asterisk from ubuntu linux. Everything is working as expected. This box is being used as a H323 gateway to the pstn. There are few complains but it is working pretty well overall. There is one thing that is bothering me. Asterisk says: May 22 05:03:39 WARNING[9360]: PRI: !! No channel map, no channel, and no ds1?

Hello I've been using VoIP over a not so reliable net: I usually get a 5% to 10% packet loss and a very high jitter. I tried several codecs and parameters, and the only thing left to test is PLC (Packet Loss Cancellement). Have the astesrisk and digium people implemented PLC?, Are they implmementing it now? and, if not, Where can i find an implementation? Thanks in advance -- Jorge

Hi i have buy two license of G729 codec, but after run Registration program i notice this error Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server, however it has created the file: /var/lib/va-infoclient which contains your machine signature and

Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this

Hi I'm buying a new box and it brings the new PCI-e standard and not the old PCI slots. I would like to know if the Digium Wildcard TDM400P and <http://www.digium.com/index.php?menu=wildcard_tdm400p2>Wildcard TE405P will work with this PCI Express. <http://www.digium.com/index.php?menu=wildcard_te405p> Has anybody worked with PCI-e yet? As far as i understand, the Wildcard

I have the following problem: I have installed two T1 digium card (old T100P cards), plus a TDM400 with 4 fxo modules. Several times in the week I have thousands of warnings like these in the log Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\uffff\uffffG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\uffff\uffffG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1

Em (20:59:21), Asterisk Users Mailing List - Non-Commercial Discussion escreveu: >On Tue, Jun 21, 2005 at 08:09:52PM -0300, Alessandro wrote: >> >> Does Somebody know why no load modules to FXS? I used zaptel-1.0.7 >> version. > >What color are the modules? >Post your /etc/zaptel.conf > >-- >Mike Mike, The color is green, red ans bluw. Link

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

zaptel compiles fine (make linux26) but when I modprobe ztdummy I get the follow error. [root@localhost zaptel]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy System info: Linux localhost.localdomain 2.6.9-5.0.5.ELsmp #1 SMP Thu May 19 19:24:44

Hi all, Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver. [root@server1 zaptel-1.0.7]# modprobe wctdm ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed

Hi All I have problem with LIBMFCR2 for once Exchange I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2. my system: Asterisk CVS 1.1.X LibMFCR2 pre.005 and unicall-pre.005 this's my setting and my setting wanpipe1.conf

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

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

Hi, Maybe someone has seen this before... I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing

I have a system that I am trying to get a port on a Sangoma A104d card connected to an Adtran 850 with 5 FXS modules and 1 FXO module. A problem I am having is figuring out what cable should be used from the port on the Sangoma to the JP2 port on the Adtran. Tried was a cross-over T1 (1->4, 2->5, 4->1, 5->2) as well as a straight T1 (1->1, 2->2, 4->4, 5->5). Neither one

I am connecting a t100p to a b8zs, superframe, loopstart t1. Previously I've attached to e&m wink and pri lines with no problems; however I seem to be missing something. Should it be fxols in the zaptel.conf (smartjack to x100p) or fxsls? With e&m wink the dnis set was 100, so it was easy to make an extension 100; and it would answer incoming calls. How do you get a loopstart to

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.