similar to: Problem with * server and FWD

Hello, I'm trying to write a Vorbis decoder (just as a hobby) but I've got a problem with reading the setup header. According to the Vorbis documentation the first data that should be read after the initial 'vorbis' string is an eight bit integer which when incremented by one, contains the number of codebooks in the header. My problem is that whenever I read this number it seems

Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

Hi everyone, I''m looking for some +1''s for a bug fix patch I submitted: http://dev.rubyonrails.org/ticket/10082 The problem is that CgiRequest#host_with_port() overrides AbstractRequest#host_with_port() and does''nt use port_string() with standard_port() care. So I created patch to rename CgiRequest#host_with_port() to host_with_port_without_standard_port_handling().

Hi all, I have an odd situation on my hands: * Two CentOS 5.2 boxes both joined to an AD domain. * Same samba version (3.0.28-1.el5_2.1) smb.conf, only the netbios names differ * Can enumerate users and groups using winbind -{u,g} on both. * nss doesn't enumerate users & groups on one (same lib versions, same conf file). //bentis@testukmcsstor1//:~$ rpm -qa | grep nss-

I am new to this list, and I searched to find the answer to my question, but could not find it. Can I do the following using "Asterisk" ... Load Asterisk on a PC running linux. Logon to VoIP service like http://www.freeworldialup.com/ to using your ethernet. Asterisk, routes the call from the PC to a regular phone connected to through the modem. When I am receiving the VOIP call, I

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to

Hi list, why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)? could anybody please give any idea to solve this issue? Please, let me know. Thanks in Advance. N.B. The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are: ***************************************

My WinXP clients are able to map a network drive on my samba server. However, it takes 30 seconds to do so, and accessing the contents of the shared drive is also quite slow. Linux (smbclient) and Windows 2000 clients are able to map the share instantly, and access files without delay. I am using Linux 2.4.5, with samba 2.2.3a compiled from source on the server. The delay is consistent at 30

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

I've tried to register * at FWD but * segfaults, so i guess my register-line in sip.conf isn't correct. It looks like: register => fwd#:pwd@192.246.69.223 should it be different? Chris

Package: logcheck-database Version: 1.2.69 Severity: wishlist Hello, when newgrp (part of the package login) is used, I see messages like this in my syslog: Aug 27 23:36:16 debian64 newgrp[1975]: user `root' (login `root' on tty1) switched to group `backup' Aug 27 19:28:15 srv1 newgrp[10082]: user `root' (login `mazur' on pts/1) switched to group `backup' Aug 27

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922

On my Ubuntu 8.04 64 bit desktop, I created an empty Rails project and from another machine, I used ab -n 10 http://210.77.27.169:3000/ to test the performance: =========================================== When I put rails 2.3.3 under vendor/rails: Server Software: WEBrick/1.3.1 Server Hostname: 210.77.27.169 Server Port: 3000 Document Path: / Document Length:

I've put a t100p in our * server and I'm having trouble configuring it. It is directly connected to an Adtran TA 750 channel bank with two FXO cards (8 analog incoming lines total). I'm able to insmod and modprobe both zaptel and wct1xxp with no trouble, but when I start * with /usb/sbin/asterisk -c I get the following output: [root@rosella root]# /usr/sbin/asterisk -c Asterisk

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX