similar to: realtime mysql db performance difference with matching extensions

Trixbox 2.2 Beta with freePBX 2.2.0rc1 I have a setup that looks something like this in ASCII art: Teliax IAX Trunk ------+ | V Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+ +--------------> Lima Office Server -----+|

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Hello All. I am stumped, please help me out.. I have the following setup: VOIP provider = VOIP GW (asterisk GW1) = VOIP server (asterisk - VS1) The gateway is there to get around the limitations running on the VOIP server. I can call out from and receive calls VS1 no problems at all. However, when I try and redirect an inbound call out via the GW, it drops out. I have found that if I

Hi, well, swift response! Qualcomm has a patched qpopper (2.5) Greetings, Jan-Philip Velders <jpv@jvelders.tn.tudelft.nl> ---------- Forwarded message ---------- Date: Mon, 29 Jun 1998 21:43:18 -0700 From: Praveen Yaramada <pyaramad@QUALCOMM.COM> To: BUGTRAQ@NETSPACE.ORG Subject: Patched Qpopper2.5 release Notification. Hello Folks, As you are already aware that qpopper

Hi Are there any known security holes or necessary precautions in using port forwarding with ipportfw? I'm planning on forwarding ports from an outer firewall/router (connected to the Internet) to a host in the DMZ, then on from the DMZ host to the inner firewall, and finally from the inner firewall to some host on the inside. Thanks, Jens jph@strengur.is From mail@mail.redhat.com Wed

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

.... does anyone onlist have any 1st-hand experience (good or bad) w/ DirectDial.com for procuring computer components ? Thanks in advance :-). I am interested in them as a potential source for parts to build an Opteron box running Linux, probably CentOS, to bring this slightly on topic .... -- William A. Mahaffey III ---------------------------------------------------------------------

Samba share access is not working for trusted domain users. I have updated krb5.conf with both domains realm and in smb.conf trusted domains default value not changed. But still share access is not working for trusted domain. # less lnxbrisglog01.02188394-667794/installed-rpms | grep samba samba-3.0.33-3.40.el5_10-x86_64 Tue 29 Sep 2015 03:25:42 PM AEST samba-common-3.0.33-3.40.el5_10-x86_64 Tue

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, I am trying to make an outbound call from phone attached to FXS port. My telephone (VoIP) line is connected to FXO port (Zap/4) Default context for channel # 4 is 'directdial' here is part of my extension.conf [directdial] ignorepat => 9 exten => 9,1,Dial,Zap/4/ exten => 9,2,Congestion include => international [international] ignorepat => 9 exten =>

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

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

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 had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

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'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' ... **