similar to: Asterisk user/host registration

Hello Everybody, I have configured a Linux box that does traffic shaping. Its working wonderfully fine, just as expected...Now i want to measure the bandwidth consumed by each of my hosts....But I dont want SNMP to run on all the hosts[as required by MRTG] I used iptraf on my linux box, but it only measures the bandwidth on interface basis only.... I tried installing traffic-vis, its not working

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

Hi, I have tried to enable IPSec support for my FreeBSD( 4.11-RELEASE) system. First, I copied the generic kernel configuration file to a file I called MYKERNEL: #cp /usr/src/sys/i386/conf/GENERIC /usr/src/sys/i386/conf/MYKERNEL Then, I added the following three lines to the options section of /usr/src/sys/i386/conf/MYKERNEL: options IPSEC options IPSEC_ESP options

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

Hi, I am using Samba 3.0.7. on a Red Hat 9 system. I tried to configure the machine as a PDC by using a configuration file quite similar to what is giving in the chapter 4.Windows NT domains of 'Using Samba' 2nd edition by Orielly. Using testparm shows that the machine has been identified as a PDC of the domain 'hamsateam.' To test the PDC, I tried to make a WinXP

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Hi, I am trying to setup ipsec tunnel between Freebsd (host1) and Linux (host2) systems.And I also interested in executing some ipsec test cases( Like TAHI conformance test suite) on the same connection. Please, suggest me some details regarding this setup and Specify any materials which can be obtained from from any locations(site).. I have enabled IPSec support for FreeBSD (4.11 Release) and

Hi All, Sorry if this is too much silly. Vorbis encoder gives out packets of information. 1. Is it that each packet contain information of one frame/window either 256 / 2048 or any length set by encoder? If yes then the encoded packet will be variable in length. 2. Every packet is divided into segments, how does the encoder decide on number of segments in each packet? It must be dependent upon

Hi, If this is true: "Granule Position Information in Ogg Header is a hint for the decoder and gives some timing and position information." So say if granule position is 10000, it means that 10000 PCM samples are encoded in this page approximately. If this is true we can neglect this information, it will not effect the decoding right(but might effect for streaming)? Ravi

hi everyone, here i have a question: is there any chance in llvm intermediate representation to compare whether a operand (ie a value) is used as a user without iterating over all the instructions in a basic block. i will explain this with an example: i=p+4; j=i+p; in the above example i mean p,4,i,p which are on the RHS are operands and the terms i,j which are on the LHS are users. so what i

Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I end up getting this when I call from 2000 to 2001. 2000, 2002, and 2001 all exist in sip.conf and I connect using them. I have all three setup to use the from-sip context. Any suggestions on what is happening? [from-sip] exten =>

I was playing today with the different queueing strategies in queues.conf when I noticed the following behavior. I have 4 agents defined in a queue in queues.conf. These agents login using AgentCallbackLogin. The strategy in the queue is set to leastrecent. I place four calls into the queue and * sends only one call to the least recently used agent. If that agent does not pick up, the

Hrm, All of a sudden for some reason Wait() and Playback() are returning non-zero and its causing calls on my inbound SIP leg not to complete. I'm not sure why -- Executing Answer("SIP/2181-4518", "") in new stack -- Executing Playback("SIP/2181-4518", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn

I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better

Sir, Given code like: > > X = add int Y, 1 > Z = mul int X, 17 while I iterate over the operands of the first instruction i want to print the variable x ,as well as the constant 1 and while i iterate over the second instruction i want to print variable x and constant 17. what should I do? thanking you, yours sincerely, anubham suresh TU-Darmstadt --- llvmdev-request at

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general] port =