similar to: About Asterisk 1.6.0.1

Displaying 20 results from an estimated 900 matches similar to: "About Asterisk 1.6.0.1"

2009 Jan 22
0
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
2015 Jul 17
2
Dovecot - Telnet error
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
2015 Jul 18
1
Dovecot - Telnet error
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.
2006 Jun 07
2
o2cb_ctl: internal logic failure while creating node
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
2015 Jul 06
1
Dovecot - Config file details
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
2016 Jan 19
3
Configure the sendmail with the dovecot.
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
2015 Jul 17
0
Dovecot - Telnet error
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. >
2015 Jul 10
1
Error while executing dovecot
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
2015 Jul 01
1
FW: Help needed to use dovecot from scratch
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
2018 Oct 02
2
Re: VM migration help required.
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, >
2010 Jan 12
2
SIP Security
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
2012 Dec 06
2
BLF and call-limit in 1.8
Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This
2015 May 11
0
"Retransmission Timeout" results in dropped calls after 32 seconds
----- 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
2009 May 14
0
Problem with Asterisk 1.4 and Linksys Spa941/962
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
2015 Dec 30
2
Signaling ringing on other extension
Ishfaq Malik <ish at pack-net.co.uk> schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default
2010 Nov 03
1
inbound call issue...
Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at
2015 May 11
2
"Retransmission Timeout" results in dropped calls after 32 seconds
----- 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
2008 Oct 14
1
SIP channels seem not to close after call is finished
Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold
2009 Apr 09
2
notifyringing=no does not work
" Hello, I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten =>
2015 May 08
2
"Retransmission Timeout" results in dropped calls after 32 seconds
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