similar to: Load Balancing

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabled by default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

Hi, I am an experienced libvirt user on Fedora versions from F15 to F17. I have developped scripts to route trafic from outside on multiple interfaces/multiples IPs to multiple VMs, and back to affect each VM the required external IP address. I have servers with more than hundreds external IPs, and up to 4 VMs, each of them route trafic on different external IPs. I have servers with Fedora

I'm trying to install the ordinal package (http://popgen.unimaas.nl/~plindsey/rlibs.html). I downloaded ordinal03.tgz and untarred it. rmutil was previously installed (and appears to work ok.) Then I installed ordinal: [root at localhost ~]# R CMD INSTALL /home/chippy/Download/ordinal * Installing to library '/usr/lib/R/library' * Installing *source* package 'ordinal' ... **

Hello, I have problems setting up HTB. This is my setup : NET 1024/256 ADSL || eth1 Linux Firewall eth0 || LAN 10.a.a.a I want to GUARANTEE for an IP (10.x.y.z) a 800kbit bandwidth for HTTP download. But When 10.x.y.z does no HTTP download, other trafic must get the whole bandwidth, of course. The script I use (see below) is quite the same as presented at (the greeeeat page) :

Hi, I was reading document http://shorewall.net/MultiISP.html#idp3634200. Inspired by the document I was trying to establish the following changes: * one additional interface: COMA_IF * COM[A,B,C]_IF interfaces request IP address via DHCP * all non-RFC 1918 destined trafic is NATed from INT_IF to COMA_IF * all non-RFC 1918 destined trafic from GW is routed via COMB_IF by default * non-RFC 1918

I have a router with about 300 clients connecting to it. It has htb with a class per client. I wnat to create a script to prioritise www trafic and ssh trafic over p2p trafic.... this is a sample of what i have now: /sbin/tc class add dev eth0 parent 1:5 classid 1:8012 htb rate 35Kbit ceil 281Kbit prio 6 /sbin/tc qdisc add dev eth0 parent 1:8012 handle 8012 sfq perturb 8 /sbin/tc filter add

Hello, does anyone know of a good calling card solution for asterisk that is able to do lcr? Does astcc do this? I've been searching around and I can find some lcr modules/apps but none that incorporate prepaid card functionality. Regards, Igor H.

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still

Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

I have had some radom occurances of someone calling in, and for whatever reason the person is getting dumped into the section of the lcr.agi file that output the message of "it is necessary to dial x". I was curious, does any one know about a variable that might be available in the AGI that would tell me what channel they are coming in on. I looked in the wiki and found

Hello all: I upgraded to 1.4.3 last night and use MySQL for CDR. I have noticed that 1.4.3 seems to log a lot of "crap" to CDR that 1.4.2 did not. I use a few macros in my dialplan to handle outgoing calls (lcr type stuff) and in addition to the proper CDR for the call itself I also have records to 's' in the same dest-context and entries to 's' in the default context.

HiI want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes.The problem is when a calls come to Asterisk1 but SIPendpoint is registered on Asterisk2.How we can check a SIP endpoint is registered or not and what is Contact?information in Dialplan ?

Hi, A friend of mine has 2 lines of 512kbps terminated in two Linux boxes. He now want to remove those 2 boxes and have some device which will loadbalance the two ISPs and also have a failover arrangement. But he has agreed to give me a chance to do it on Linux for my own satisfication. Is this easy to do with lartc? How do I go about it exactly? I have very less time to do it since his

Hello All, Right now I have several providers. Voipjet, Teliax, and more recently Broadvoice. Broadvoice gives me unlimited to europe, but what I want to do is determine the best way to setup a dialplan so for example, certain countries will go through the cheapest route. I am really only interested in Poland, Russia and Turkey. Poland is free on broadvoice, but not for cellular, which I may