similar to: Asterisk Can't Run

Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04:

I've just installed a Digitnetworks X100P clone in my * server and run the install script for the voicepet-single-x100p tarball. The install appeared to run OK with modprobe wcfxo successful and the ztcfg reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)". When I try to start * though I get a segmentation fault after loading res_features.so. I discovered that the

I am putting together my first *. I had it running with two other pc's running xlite and setup voicemail and a couple of menus and submenus and had that running well. I had order a couple of oem x100p cards from digitnetworks. I installed them as they said with their voicepet2.2.zip drivers and did the modprobe on zaptel and wcfxo and then ran ztcfg -vv and got this: Zaptel Configuration

Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

I appreciate any recomendation of a simple CallingCard Application and resources of users manual. __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

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Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. "chan_iax2.c:5550 socket_read: Call rejected by ****: No authority found" iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that

I installed X100P from DigitNetworks. The system found the Wildcard X101P and i was able to modprobe zaptel and wcfxo and ztcfg. then, after i compile asterisk, i am able to make third party voip calls dialing from the asterisk cl and using a regular headset. but when i try to dial out from zap channel, i don't have a dial tone and i hear the same click when i make voip calls. when i dial a

Hi: The phone company here use Europe telephone system. They offere ISDN lines with multiple channels, starting with two and they add channels on demand. Am I write to understand ISDN of more like E1 or T1 line. If not, what's the difference. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

Hi: I want to connect a wi-fi phone to my Asterisk box through a wi-fi AP so I can make voip calls. please send me your recomendation about what wi-fi phone I should be looking for. Anybody tried the HOP1502 Wi-Fi IP phone. Its listed price $39. Regards; Chawki ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

Hi everybody I have a problem and want to know if anyone has already seen it before : I try to use web-meetme.3.1.0 and follow these instructions http://sourceforge.net/docman/display_doc.php?docid=48924&group_id=164788 1) when i do "make" command in cbmysql folder, errors happened ********************* cc -pipe -I/usr/include/mysql -L/usr/lib/mysql -fPIC -I/usr/src/asterisk

There was errors when I tried to start the script recommended by Andrew to boost bandwidth for voip http://www.mixdown.ca/~andrew/dump/rc.tc. This is the output I get : ./rc.tc start RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File

Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for

Hello, I have upgraded to 1.0.4 version of asterisk. After that asterisk crash every time On receiving an call from iax2 trunk to musiconhold application. SIP calls to MusicOnHold is however working. I already upgraded to 1.0.5, but the problem still Remainig. Any idea ? Iax2 : call proceding : Jan 25 17:29:40 DEBUG[9997]: pbx.c:1261 pbx_extension_helper: Launching 'WaitMusicOnHold'

I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales

Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored...

Hello Folks, I'm trying to decide here between a few cards for connecting an Asterisk box to a single E1 channel (either PRI or R2 signaling): - Digium E100P: has been replaced by the TE110P below, but can still be had at places like digitnetworks.com for $475, and I guess there's always a place for good-olde-obsolete cards in the world as long as they work :-) - Digium TE110P: