similar to: Asterisk Message Logs

Hello, I have two user machines, each with a jain-sip-applet-phone installed on it. I use the following process to try to make a call: 1. Register each phone with the Asterisk server (working). 2. Add a contact in each phone which is the other user. (Get a "489 Bad Event" SIP error shown below in red) 202 at 192.168.1.252 has been added to your contacts. null send request:

Hello, I have been reading up on the capabilities of the Asterisk-Java library. I believe that this library can act as an interface between a Java GUI(custom softphone) and the Asterisk server. Seems like the Live API would be easiest to use to make the connection to the Asterisk server and to set-up calls. One thing I am not sure about is how to transmit the audio streams between users'

Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: "Oooh, format changed to 2". Would anyone know why

Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important

Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would

Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a "401 Unauthorized" error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ;

Hello, I have the Jain-Sip-Applet-Phone installed on two machines in a small LAN network. These machines are connected through an Asterisk Server (Using Trixbox). I run the phone as an application on both machines through Eclipse and I am able to log on as a user with one of the extensions that I use within Asterisk on each machine (extensions 201 and 202 in this case). When I try to add a

Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable

Hello, I was wondering if it is possible within Asterisk to be in many meetme conferences at the same time. This would be sort of broadcasting over all the conferences at once. Thanks, Denis

Hello, I have a small LAN network connected through an Asterisk Server (Trixbox). I was looking to create my own custom made softphones, and I have been looking into how to transmit and receive via RTP. Would anyone know how the Asterisk RTP bridging works, and if there is any documentation on it? How is the RTP stream routed through the Asterisk server? Do I just give it the endpoints and

I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason

On 12/19/2017 03:57 PM, Manish Jain wrote: > > > On 12/20/17 01:45, m.roth at 5-cent.us wrote: >> Manish Jain wrote: >>> >>> On 12/19/17 22:11, Manish Jain wrote: >>>> On 12/19/17 22:07, Jonathan Billings wrote: >>>>> On Tue, Dec 19, 2017 at 02:54:36PM +0000, Manish Jain wrote: >>>>>> I uninstalled the old TV, and

On 12/19/2017 02:57 PM, Manish Jain wrote: > > > On 12/20/17 01:45, m.roth at 5-cent.us wrote: >> Manish Jain wrote: >>> >>> On 12/19/17 22:11, Manish Jain wrote: >>>> On 12/19/17 22:07, Jonathan Billings wrote: >>>>> On Tue, Dec 19, 2017 at 02:54:36PM +0000, Manish Jain wrote: >>>>>> I uninstalled the old TV, and

It appears that part of the problem could be that when logger rolls over the mongrel log file. There is a mutex synchronization issue where the various mongrel instances all try to write to the new log file and block each other. This keeps them in a continual Application 500 Error state. Has anyone run into this problem? With log rotation turned off the problem does not appear. - Jared Brown

with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI> module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from

Manish Jain wrote: > > On 12/19/17 22:11, Manish Jain wrote: >> On 12/19/17 22:07, Jonathan Billings wrote: >>> On Tue, Dec 19, 2017 at 02:54:36PM +0000, Manish Jain wrote: >>>> I uninstalled the old TV, and installed the version you indicated, but >>>> I >>>> get nothing at all: >>>> >>>> /home/bourne # teamviewer

Darrick J. Wong would like to nominate Anand Jain to help more on btrfs testing part (tests/btrfs and common/btrfs). He would like to be a co-maintainer of btrfs part, will help to review and test fstests btrfs related patches, and I might merge from him if there's big patchset. So CC him besides send to fstests@ list, when you have a btrfs fstests patch. Signed-off-by: Zorro Lang <zlang

Signed-off-by: Anand Jain <anand.jain@oracle.com> --- mkfs.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/mkfs.c b/mkfs.c index de1beed..0843600 100644 --- a/mkfs.c +++ b/mkfs.c @@ -1196,8 +1196,7 @@ static void process_fs_features(u64 flags) for (i = 0; i < ARRAY_SIZE(mkfs_features); i++) { if (flags & mkfs_features[i].flag) { - fprintf(stderr,

Remote work possible. Location: Columbia, SC Duration: 12+ months Rate: $60/hr Daily Responsibilities: Ruby coding, using rails, testing, generating ad hoc reports for client as needed, providing daily status updates to team, Presence on internal IM system during work hours. Required Skills : RUBY, RAILS, GIT, MYSQL, CUCUMBER, RSPEC, JQUERY, EXCELLENT, LINUX, OS X, JSON, COMMAND

On 12/19/17 19:55, Johnny Hughes wrote: > On 12/19/2017 07:16 AM, Manish Jain wrote: >> Hi, >> >> I am writing a book about Unix (Linux / FreeBSD) and my book needs >> documentation for TeamViewer13 running under Centos 7 (64-bit). >> >> Unfortunately, running TeamViewer does not get me a GUI -- the install >> went smoothly and teamviewerd daemon is