similar to: Query About Asterisk 1.6.0.1 Dialog Event Package.

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Hello All, I have installed dovecot and started it. However when I try to telnet getting below error means dovecot starting is not successful. telnet localhost 10110 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Connection to localhost closed by foreign host. Please advise. Thanks and Regards, Sachin Kulkarni

It was indeed successful. You should be worried if you don't get the message. On 17 Jul 2015 20:01, "Shane James" <shane at phpboy.co.za> wrote: > Port 110 is what you want > > > On 17 Jul 2015, at 17:43, Sachin Kulkarni1 <SK00335601 at TechMahindra.com> > wrote: > > > > Hello All, > > > > I have installed dovecot and started it.

Hi, I am getting following tyoe of error while trying to configure node using ocfs2 o2cb_ctl: internal logic failure while creating node My Kernel version is 2.6.9-22.ELsmp Rpm which i use are ocfs2-2.6.9-22.ELsmp-1.0.7-1.i686.rpm ocfs2console-1.0.3-1.i386.rpm ocfs2-tools-1.0.3-1.i386.rpm after that I gave command ocfs2console Select Cluster ? Configure Nodes Click on Add and enter the Name

Hello All, We want to install / configure and test Dovecot. We are doing it for the first time. After installation we are now looking for configuration file and need to know significance of below params and what values we should provide ? protocols = listen = base_dir = login_greeting = Please help. Thanks and Regards, Sachin Kulkarni

Hi Dovecot team/All, First of all, I always appreciated your contribution & effort. Can you please provide the step for configure sendmail with dovecot. We could not understand the parameter which mention in sendmail.cf file. Please provide the parameter description as well as where need to change exactly in sendmail.cf file. Thanks and Regards, Lucky

Port 110 is what you want > On 17 Jul 2015, at 17:43, Sachin Kulkarni1 <SK00335601 at TechMahindra.com> wrote: > > Hello All, > > I have installed dovecot and started it. > > However when I try to telnet getting below error means dovecot starting is not successful. > > telnet localhost 10110 > Trying 127.0.0.1... > Connected to localhost. >

Hello All, I am trying to execute dovecot by below command /opt/app/dovecot/latest/sbin/dovecot -c /opt/app/dovecot/latest/etc/dovecot/dovecot.conf I am getting below error doveconf: Fatal: Error in configuration file /opt/app/dovecot/latest/etc/dovecot/dovecot.conf: default_login_user doesn't exist: dovenull please guide me to resolve this. Thanks and Regards, Sachin

Hello All, Can anybody help in this please? Thanks and Regards, Sachin Kulkarni -----Original Message----- From: Sachin Kulkarni1 Sent: Tuesday, June 30, 2015 6:13 PM To: dovecot at dovecot.org Cc: 'dovecot-owner at dovecot.org'; 'dovecot-request at dovecot.org' Subject: Help needed to use dovecot from scratch Hello, I need to use Dovecot. So need to install it from scratch

On Mon, Oct 1, 2018 at 12:25 PM Nitin Rane <Nitin.Rane@techmahindra.com> wrote: > Hi Arik,Shahar, > (Adding libguestfs mailing-list) Hi Nitin, > I am looking to migrate VMs from Openstack Juno to Openstack Newton and > from VMWare to Openstack Newton. > > While doing google search I came across the link authored by you, >

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:24:53 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > Could this

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:35:07 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > That should

Hello, I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 4:18:58 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > ----- Original

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Friday, May 8, 2015 5:12:28 PM > Subject: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Hello, > > I am

Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the

>Also, I'm sure you've probably checked on this one, >but are the phones registered with asterisk? >You can make outbound calls on them without them >actually being registered. I'm assuming you can >still get in and see the CLI. What does "sip show peers" >look like? What does "sip show peer xxx" show? >What does the CLI show when you

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Monday, May 11, 2015 12:32:06 PM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32

I am using PBXware for configuring users and extensions. Pbxware uses Internal script called init.sh to process the calls based on its own version of extensions.conf defined in the GUI. I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. I have used IAX2 extension 101 and dialed SIP Extension 51 But the PBXWare's Init.sh AGI command identifies the DNIS as another IAX