similar to: How to set-up LCR

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

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

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

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

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing

Hello, How do you guys implement LCR in Asterisk? Thanks,

Hi. Thank you to Olle, the Wiki, and Trollphone. I found the Trollphone Rate Engine listed on the Wiki at: http://www.voip-info.org/wiki-Asterisk+addon+rate-engine I'm not familiar with LCR yet, but this is something that I need to do. I have it all installed but am not familiar with the terminology. Can someone help me maybe figure out what some of these fields are for? mysql>

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

Hello, I short question: I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR. Do I need chan_lcr? I have: Asterisk 1.8 mISDN .v2 integrated in Kernel 3.0.22 mISDNuser lcr 1.7 HFC-E1 Evaluation board from cologne chip I tried to configure Asterisk with <./configure --prefix=/usr/src/lcr --with-gsm-bs> and it runs without errors, But make and make install didn't run

I have got Areskicc installed with AMP and I can't stress out how good and excellent this software is. I must say the author desire every credit for this software and I would like to say Thank you to Arezqui who wrote the software. Now there is one last thing before my whole asterisk calling card become prefect. And I am wondering whether anybody may know the answer of this. I

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 guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i

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, This is what I want to do: 1. Asterisk to answer calls via DID's, currently using SIPGATE 2. Provide a menu, and allow users to dial out. 3. According to the country and area they dial, the call should connect via one of up 4 carriers depending on cost. 4. If the carrier is busy it should go to the next one in line and so forth. I have tried to set this up, but it never answers the

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' ... **

I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I

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.

Hiyall. Just wondering how people test your emergency dialing in the UK. Obviously you need to dial the 999 for emergency services, but am a bit unsure if this would go down too well with the operator with a 'sorry just testing' call. (you do all /test/ your emergency dialing dont you!?:-) ) As another thing, what is the correct method when using least cost routing... If you have a

Hello everyone! I am using wine 0.9.30 with openSUSE 10.2 I've tried to install and run VoipStunt, and program installs with no error, but fails to start with the following output: dodo@Locutus:~> wine "C:\Program Files\VoipStunt.com\VoipStunt \VoipStunt.exe" preloader: Warning: failed to reserve range 00000000-60000000 err:module:import_dll Library gdiplus.dll (which is needed