similar to: Call forwarding in Asterisk

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to

Hello List, Good day, We have faced an issue in Asterisk, the issue is we are not able to make manager connections and we could see two pid's for the service asterisk status command. The sequence of operations in our setup goes here, 1) Linux machine is rebooting , we have issued a service asterisk restart command. 2) Tried to connect from an application to asterisk.

I am facing one problem with rsync. in /etc/rsyncd.conf file , one of my entry is like bellow. path = /data/Knowledge Centre With this, rsync report an error "chdir failed, no such directory". Even i tried like bellow path = /data/Knowledge\ Centre with this also, it gives the same error ? How to overcome this ? what is the right syntax ? -- Thanks & Regards, Shashi Kanth

Hello Rowland, I do not have support contract with RedHat, and due to some application dependency i have to be on 5.8. No choice for me to upgrade the OS. I have choice to upgrade Samba from 3.5 to 3.6.6 through RPMs, but i am not really sure whether it solves my core issue. Coming back to my original query " CIFS Null Session vulnerability ", just i would like to understand whether

Hi Volker, Reaching to RedHat is not the option for me right now, could you please tell me are there any configuration parameters in smb.conf which will avoid issuing NULL or Anonymous connections or sessions. On Thu, Apr 26, 2018 at 12:49 AM, Volker Lendecke <Volker.Lendecke at sernet.de > wrote: > On Wed, Apr 25, 2018 at 11:12:07PM +0530, Shashi Kanth Boddula via samba > wrote:

Hi Volker, I am not finding anywhere the Samba 4.X RPMs for RHEL 5.X platform. Please share if you know any place from where i can download. I am afraid to build from source code. On Thu, Apr 26, 2018 at 11:01 AM, Volker Lendecke <Volker.Lendecke at sernet.de > wrote: > On Thu, Apr 26, 2018 at 04:21:39AM +0530, Shashi Kanth Boddula wrote: > > Reaching to RedHat is not the option

Hi Everyone, I have Samba server 3.5.10 running on RHEL 5.8 platform and it has joined to our AD domain controller. Recently my Windows guys has done some changes to AD Security by stating " CIFS Null Session Vulnerability Fix via GPO - Security Requirement". After this change, my windows clients are not authenticating with domain credentials while accessing the shares, but nothing has

Hi, In my company , we have 25 HP printers at different floors. All printesr are configured on One Linux Machine. Each printer has one administrator. My task is share the printer to that particular administrator. For example if i have 2 printers ( HP1 and HP2) , i want to share HP1 printer to only 192.168.0.1 and HP2 printer to only 192.168.0.2. If we use host allow &

Hello List, Good day, We have an application, where we redirect a channel to meet me. Sometimes the channel is getting hanged up by Asterisk, and we get an hang-up event. Please reply back, if any one faced such issue. Here is the hangup event info, -HANGUP {calleridname=<unknown>, connectedlinename=<unknown>, uniqueid=1384413814.79523, cause=0,

-----BEGIN PGP SIGNED MESSAGE----- ============================================================================= FreeBSD-SA-02:20 Security Advisory FreeBSD, Inc. Topic: syncache/syncookies denial of service Category: core Module: net Announced: 2002-04-16

I have several large customers (200+ extensions) running on vSphere without issue. Not sure about OpenVZ, thought. 2015-04-07 11:36 GMT-03:00 Mitul Limbani <mitul at enterux.in>: > Show him this freaking thread, or else ask him to prove it otherwise. > > We all here have decades of exp dealing with asterisk. > > Mitul > On 07-Apr-2015 7:27 PM, "Ikka

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Could you explain more ? One people can access to one printer or each printer are one adminitrator and other people can just print ? else there are the printer admin parameter ----------------------------------- St?phane PURNELLE stephane.purnelle@corman.be Service Informatique Corman S.A. Tel : 00 32 087/342467

Hello Sir, I have few clarifications as we are planning to work with asterisk. If you don't mind, please clarify the following:- Q1. Do Asterisk support ACD functionality? If Yes, can you give information on how to configure or work with ACD (and it's usage). Q2. From the list of features listed in www.asterisk.org , I see "Predictive dialler" is listed

I'm having the same issue! The difference in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00

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