similar to: session border controller

Hi ? I ?wonder if?anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT? - NAT - Server Client can hear users from server side but server cant hear client. ? Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no ? Ive check the sip headers in the debug mode and its using the external address in

Hi ? I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming

I am curious if it is advisable to use implement Asterisk as a Session Border Controller for a VoIP reseller environment. Users will terminate calls SIP to my server, which will authenticate them via RADIUS, perform a LCR lookup, select an appropriate trunk (based on LCR), and terminate the call (update RADIUS accounting at end of call). All while acting as a B2BUA to prevent the users from seeing

Come on you carriers on the list... Give up the dibs what are you using and why? About to sledgehammer these SELECT * FROM GARBAGE WHERE SBC = 'nCite' Don't bother shooting me off Newport Networks stuff... Too pricey -- ==================================================== J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 echo infiltrated.net|sed

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

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

@Jonathan Goodwin: Do I have to use a flag to tell it where the LLVM include files are? And would it be easier to do that in the CMake GUI? Please tell me how you had it find the include files if that may help me. ________________________________ From: Jonathan Goodwin <jondgoodwin at gmail.com> Sent: Wednesday, November 28, 2018 6:24 AM To: zturner at google.com Cc: Osman Zakir; llvm-dev

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

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

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

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

I don't know what executable to point it to for SVN because, like I said before, I just right click where I want to check something out and click what I need in the context menu. So I need to know what executable to point it to. ________________________________ From: Osman Zakir <osmanzakir90 at hotmail.com> Sent: Thursday, November 29, 2018 11:06 PM To: Zachary Turner; llvm-dev at

hello there. i do have debian 6 in alix 2d13 that runs asterisk 11.7. i do need to interface an ISDN line to asterisk and i bought an external isdn card usb. it is a dlink du 128ta. i have seen that i need misdn and lcr. in a clean install of debian 6.0.8 i installed through git the misdn and misdn user. unofortunatelly i am stuck with the lcr. i have installed misdn, misdnuser and LCR through git

I think most people are using ninja to do the build. You can still use the Visual Studio generator for the IDE experience while doing build from ninja. I know there are a couple of people who build from inside of Visual Studio, but it's a pretty small number, and I don't think they are targeting WebAssembly. On Tue, Nov 27, 2018 at 12:46 PM Osman Zakir via llvm-dev < llvm-dev at

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

I need them so I can build stuff using clang or clang-cl with its C++ standard libraries. To make sure that lld-link won't give errors about missing symbols from any standard libraries. By the way, you wouldn't happen to know how to use link.exe, would you? I might need some help on that to understand how to use lld-link.exe. ________________________________ From: Zachary Turner

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

Yeah, I don't think that matters. It did set the toolset architecture to x64. You know, I attached those log files for a reason. ________________________________ From: Zachary Turner <zturner at google.com> Sent: Tuesday, November 27, 2018 4:27 AM To: Osman Zakir Cc: llvm-dev at lists.llvm.org Subject: Re: [llvm-dev] Have LLD and Clang in their correct locations, but still can't

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' ... **

Hi All, I'm testing different ways to implement a LCR/OCN tabe to shift calls to multiple carriers for better rates. I'm using realtime mysql database access, asterisk 1.2.9.1 with mysql 3.23. Scenario 1: I send outgoing calls with with a Goto statement into the context with the realtime switch to dip into the mysql lcr database, currently I have ~13K records/routes. The database