similar to: Get SIP to work over very limited network access

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hi, Has anybody seen a non commercial, or freeware, or GPL, or even CHEAP... POP/IMAP to Text-to-speech? I have a working version for POP3 using festival. It DOES work... it even cleans the email contents to get the actual content. It works great with Outlook emails and similar, and skips non multipart/alternative (that would be mainly SPAM, where the email is just

Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James

I'm looking for a way to transmit a user specific variable to my dialplan If we use the example of the hair color, I was thinking of having something like: [bob] context=users host=dynamic secret=password type=friend username=bob hair=brown [lary] context=users host=dynamic secret=password type=friend username=lary hair=black And be able to access from the dialplan: [users] Exten =>

I've a data frame with this structure: 'data.frame': 1987 obs. of 11 variables: $ site : Factor w/ 24 levels "B(W)","BC-1",..: 1 1 2 2 2 1 1 1 ... $ sampdate : Date, format: "2000-07-18" "2000-07-18" ... $ tclass : Factor w/ 8 levels "Annelida","Arachnida",..: 1 5 5 5 5 ... $ torder : Factor

Hi, I'm setting up multiple asterisk servers and trying to do the classic DIAL(IAX2/asterisk1/${EXTEN}&IAX2/asterisk2/${EXTEN}&IAX2/asterisk3/${EXTEN},15) After googling a bit, I fell on a discussion about putting this in a variable so that adding additionnal servers would be easy. I can't seem to find the link anymore, but it went something like this: extensions.conf:

Hi everyone, I'm having a little problem and was wondering whether anyone would have any ideas or pointers for me. I've been working on load-balancing asterisk and have had a pretty successful setup using LVS and IP tunneling (plus a bit of iptables nating). I am only load balancing the SIP registration while the RTP between the SIP phone and the asterisk server and between the

Hi everyone, I just got myself a IAXy device and am trying to integrate it to our asterisk server. I configured the IAXy and it is registering and I get a dial-tone. If I try calling another SIP device, and I get "can't translate IAX2 to SIP" How can I make my IAX device communicate with a SIP device (and vice-versa)? Here's what the log says: -- Executing

Hi all, I am fairly new to R and I am trying to run mvpart and create a MRT using explanatory variables and covariables. I've been following the procedures in Numerical Ecoogy with R. The command (no covariables) which works fine - ABUNDTMRT <- mvpart(abundance ~ .,factors,margin=0.08,cp=0,xv="1se",xval=nrow(abundance),xvmult=100,which=4) where abundance is 4th root

Hello all, I´ve read http://lartc.org/howto/ and now i am just confused, i think my skills with linux are not very good, so asking for help. I have a linux box with two ethernets cards eth0(gateway 1mb) with is the host for some sites and emails and eth1(nat interface) with provide internet acess to other 5 pcs. I would like to limit the bandwith 512 k for the eth0 and 512 k for eth1 however

Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan andytan@fastmail.fm -- http://www.fastmail.fm - Does exactly what it says on the tin

Hello, I''m Alexandra Well, i have been working with cbq around 3 month ago without any problems, what i do is control the bandwith to 64Kbit for input and output and it works well. But now I don''t want control the Bandwith, i only need prioritize udp trafic over tcp trafic Can somebody to help me with my requirement? Thanks Alexandra

Hi, i''ve following problem. One of our gateway router, which connects some of our customers should have bandwith limit. So customer A with IP XX should have 2 Mbit, customer B with IP YY should have 10 Mbit. There is no need of borrowing bandwith so no fairness needed. My simple question: with which technique should I manage this shaping? Or is there any existing project which

Has anyone had any experience using a VOIP provider in Japan? No matter what I try, my REGISTER string kicks back one of 2 errors: Got SIP response 481 "Call/Transaction Does Not Exist" back from x.x.x.x or Got SIP response 400 "Bad Request" back from x.x.x.x My register string is as follows: 05075034132@ipphone2.freebit.ne.jp I have tried the following also:

hi, (assume HTB) I was thinking will it be possible to do some sort of UNFAIR-SHARING :"), what I have in mind : Say I have a internet link with 100kbits bandwith, then I want to share it between many clients (which will increase over time). Let''s i start with 5 clients with rate = 30kbits... See the total bandwith of users is 120kbits but I have only 100kbits.... So where is the

Hello List, I''m new to QoS/tcng/HTB and friends, so please forgive me if my question might be silly... After having read lots of HowTo documents I''m totally confused... The Challenge: ============== I''ll have to deploy several "mirror" download servers (Linux) which must be able to handle a huge number of HTTP download requests (about 10k to 20k unicast

I would like to know how to calculate the amount of bandwith I would need to host X number of calls. For example, if user A in San Francisco with an ATA 186 calls user B in New York with an ATA 186 and Asterisk is being hosted in a PC in Miami. How much bandwith do I need to have in Miami? Do I just need bandwith for the setup of the call (ie the SIP part) or are there any instances where the

I have clients, which connectin to Internet through vpn. I want to dynamically share bandwith between vpn connections, so if there few connections, then they get all bandwith, if more then they get their minimal guaranteed bandwith. my idea is: ip-up.local: tc class add dev $DEV parent 1:1 classid 1:2${1/ppp/} htb rate $[$RATEUP/$VPNS]kbit ceil ${RATEUP}kbps tc filter add dev $DEV protocol ip