similar to: Load module chan_zap.so failed

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

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

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

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 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

Is there is an open source ACD component for Asterisk? Likewise, is there an open source CTI component that will work with Asterisk? Regards, Jim O'Brien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/e6ee98da/attachment.htm

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

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?

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

chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks

I'm running latest asterisk and zaptel, I have loaded wctdm and lsmod shows that it is in the kernel. I have configured the FXS and FXO ports on my TDM400P, and ztcfg shows both as configured with no errors. When I start asterisk I get the following error: Mar 13 14:07:41 WARNING[10958]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_cal. A

Hi everybody, I am working with gams and I have found some questions when plotting gams models. I am using mgcv, and my model looks something like this: model<- gam(x ~ s(lat,long)) I can plot the output of the model using plot(model) or plot.gam(model) and I get a surface plot. That is ok, but what I want to do now is to extract the data used to perform the surface plot. Like that I

A customer contacted us today concerning getting a VoIP to PSTN system with a few IP Phones setup. Asterisk should fit his needs. It is not a big job, but I think that this customer is going to need onsite work. Please contact me off list if you are an interested reseller in the Washington, DC area. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389

Hello, I have researched a few postings where users mentioned being able to install Asterisk on Mac OS X Panther by adding some code after line 165 in the Makefile and then compiling. This has been unsuccessful for me. I downloaded the asterisk-0.9.0.tar.gz tarball and am trying to install from it. The output I get upon trying to make after editing the Makefile can be viewed at:

I have linux box (does nat and firewall for small network) connected to dsl. I want to set priorities for protocols (that nothing could disturb web browsing). This is my rules (eth0 connected to internet): /sbin/tc qdisc del dev eth0 root /sbin/tc qdisc add dev eth0 root handle 1 htb default 30 r2q 100 /sbin/tc class add dev eth0 parent 1: classid 1:2 htb rate 900Kbit burst 15k /sbin/tc class

I have a couple of questions I'm hoping folks can help me out with. When I search the mailing list, I see folks doing what I'm interested in so here's hoping ! - How are people making out with interfacing to the 7960? I'm considering buying a number of these as they look quite feature rich. But, are they easy to interface to? - Will I be able to interface with softkeys on

I'm having problems with a new install of Asterisk (I had to reinstall because hard drive failed). I've used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for

Hi, I've connected a two T100P from digium with a 2 Rhino channelBank. Everything is working as expected. but I have occasional Falls, asterisk take 99% of CPU resources, with the following report May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n**

Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227