similar to: Trying to find good VOIP provider.

Hi, folks. I am trying to implement reverse proxy using squid with mac_portacl, but i have problem while binding squid to port 80. Am i missed something? Here is my mac_portacl variables: # sysctl security.mac.portacl. security.mac.portacl.enabled: 1 security.mac.portacl.suser_exempt: 1 security.mac.portacl.autoport_exempt: 1 security.mac.portacl.port_high: 1023 security.mac.portacl.rules:

RSpec-1.1.3 has been released. If you''re using RSpec and autotest, you''ll have to upgrade to RSpec-1.1.3 and ZenTest-3.9.0 at the same time. == Version 1.1.3 Maintenance release. * Tightened up exceptions list in autotest/rails_spec. Closes #264. * Applied patch from Ryan Davis for ZenTest-3.9.0 compatibility * Applied patch from Kero to add step_upcoming to story listeners.

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

Hi all, I found similar sequences in the /var/auth.log files of freebsd boxes, I supervise.: Aug 13 13:56:08 www sshd[26091]: Illegal user test from 165.21.103.20 Aug 13 13:56:11 www sshd[26093]: Illegal user guest from 165.21.103.20 Aug 13 13:56:15 www sshd[26096]: Illegal user admin from 165.21.103.20 Aug 13 13:56:18 www sshd[26103]: Illegal user admin from 165.21.103.20 Aug 13 13:56:21 www

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, folks. I want to buy FXS-SIP terminal with 4 ports (up to 250$). Do you have any recomendations and Asterisk configurations samples for such devices. Any pitfalls? Actually i realy don't know what to buy? -- ========================================================================= = Best regards, Nikolay Pavlov. <<<------------------------------------ =

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

Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

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

Running freeBSD 6.1 After changing chkrootkit to the latest version V. 0.47 and compiling it then running it I get the following: ==================<SNIPPIT>================ Searching for anomalies in shell history files... nothing found Checking `asp'... not infected Checking `bindshell'... INFECTED (PORTS: 6667) Checking `lkm'... You have 131 process hidden for readdir

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

Anybody using Cisco Call Manager and connecting to any SIP termination service like voipjet, voxee, etc? Please msg me offlist. AK

Hi, all I need building my own install server and want install my custom world, custom kernel and other in non-interactive mode. With 6.1-beta3 it work in mass (I not try install my kernel). With 6.1-beta4 and more fresh I can't get non-interactive installation. 1. I can't find how to select timezone non-interactive. 2. I can't good variant for installing my own kernel (now I use

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 all, I am using a headset and the X-lite softphone (sometimes I use IAXComm, but I'm having difficulties using OSS emulation with it) to connect via uLaw to my internal Asterisk server, which is a Pentium II 400 with 128 megs of RAM. After getting this headset, most or all of the echo people on the other line were complaining about is now gone, according to them. However, every

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Seems there were a couple of correlated failures that appear to be flakes on this buildbot recently: green: http://lab.llvm.org:8011/builders/clang-ppc64le-linux-multistage/builds/13974 red: http://lab.llvm.org:8011/builders/clang-ppc64le-linux-multistage/builds/13975 (target-override.c during stage 1, seems to be missing the directory/symlink it just created) red: