similar to: Asterisk 1.2.18 problem

Hello all, have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a ".gsm" audio file to use it as a voicemail file with Asterisk. Thanks. Abdelkader Mosbah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070812/2ffc7135/attachment.htm

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

Hello, I have the following situation: A firewall is blocking all SIP and RTP traffic in the side of some of my clients. My clients cannot change settings of the firewall. I need to solve this problem and I need some help from you. I have this idea: implement a SIP user agent which does not use well known SIP ports (uses http port 80 for example) and use other ports that are not blocked

Hello All, MOS and R factor are the two QoS parameters used to estimate VoIP call quality. I have found that they are calculated from other metrics like jitter, latency, packet loss,...etc. But, haven't found any formula or arithmetic rule to calculate them. Do you have an idea about their formulas or an open source that calculates them. Is it possible to interpret them from wireshark.

An attacker is scanning my Asterisk Switch to gain illegitimate access to VoIP call functionality. Using a sip scanning tool, *it* sends REGISTERs with random identities. And when it discovers one identity subscribed in my switch, it tries to authenticate with random passwords using this user name. For the moment, I have replaced this account. And also blocked the IP it has used but each time

Hello all, After installing Asterisk, i have installed the docs by "make progdocs". But i don't know where to locate this documentation. please Help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm

The failregex statement in my jail.conf file is: * failregex* = NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Wrong password NOTICE.* .*: Registration from '.*' failed for '<HOST>' - No matching peer found NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Username/auth name mismatch

What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of MOSBAH ABDELKADER Sent: Saturday, August 04, 2007 3:16 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Connecting

Hello, What is the maximum number of simultaneous calls supported by asterisk. thks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090703/0794c554/attachment.htm

Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 -----------------> connected to -----------------> FXO interface in PBX2 =============>

Hello, Did anyone succeeded in installing Asterisk on OpenWRT system. pls help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090724/aee7ee12/attachment.htm

Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at

I've found OpenVPN to be easy to configure and very robust. It has a zillion options, but they are just that - options. I haven't used it for VoIP, but I've put it to good use doing layer 2 bridging which has eliminated many problems with certain programs traversing NAT and load-balancing routers. I can't think of any reason why it would not work well with Asterisk. > On

Hello, R users, I have two factors (treat, section) anova design experiment where there are 3 replicates. The objective of the experiment is to test if there is significant difference of yield between top (section 9 to 11) and bottom (section 9 to 11) of the fruit tree under treatment. I found that there are interaction between two factors. I wonder if I can contrast means from levels of

I am confused whether Student's sleep data "show the effect of two soporific drugs" or Control against Treatment (one drug). The reason is the next: > require(stats) > data(sleep) > attach(sleep) > extra[group==1] numeric(0) > group [1] Ctl Ctl Ctl Ctl Ctl Ctl Ctl Ctl Ctl Ctl Trt Trt Trt Trt Trt Trt Trt Trt Trt [20] Trt Levels: Ctl Trt > sleep$group [1] 1 1 1 1 1

On Thu, Sep 06, 2018 at 12:05:23PM +0800, Jason Wang wrote: > This patch introduces to a new tun/tap specific msg_control: > > #define TUN_MSG_UBUF 1 > #define TUN_MSG_PTR 2 > struct tun_msg_ctl { > int type; > void *ptr; > }; > > This allows us to pass different kinds of msg_control through > sendmsg(). The first supported type is ubuf

On Thu, Sep 06, 2018 at 12:05:25PM +0800, Jason Wang wrote: > This patch implement TUN_MSG_PTR msg_control type. This type allows > the caller to pass an array of XDP buffs to tuntap through ptr field > of the tun_msg_control. Tap will build skb through those XDP buffers. > > This will avoid lots of indirect calls thus improves the icache > utilization and allows to do XDP

Hi! In R 3.0.0 from current SVN, ?summary.lm says: > Value [...] > df degrees of freedom, a 3-vector (p, n-p, p*), the last > being the number of non-aliased coefficients. ?summary.glm says: > df a 3-vector of the rank of the model and the number of residual > degrees of freedom, plus number of non-aliased coefficients. It seems to me that the description is reversed: p is

Wim, Below are two problems I found in testing the newly released ocfs version. Just for your information. Gr, Robert. - Copying of multiple files gives errors [oracle@prac01 test]$ cp --o_direct -R ./a2/* ./backup/a2/. cp: writing `./backup/a2/./ccdata.dbf.bck': Invalid argument cp: writing `./backup/a2/./ccindex.dbf': Invalid argument cp: writing `./backup/a2/./control02.ctl':

Hi All, With the help of Chuck Tuffli, I'm now able to use CTL to export a zvol over FC to a Linux host: LUN Backend Size (Blocks) BS Serial Number Device ID 0 block 4185915392 512 FBSDZFS001 ORA_ASM_01 lun_type=0 num_threads=14 file=/dev/zvol/tank/oracle_asm_01 1 block 4185915392 512 FBSDZFS002 ORA_ASM_02