similar to: Softphone RTP Session Start-up Delay

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'

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 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

Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through

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 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

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

Dear all I have setup of asterisk 1.4.11 Now i want soft phone which one support file sharring + video + voice call with asterisk SIP is there any soft phone which support this all feature ?? with asterisk Regards Satish Patel --------------------------------- Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.

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

Hi all! Im looking for 1ezphone to use as a web softphone but I?cant access to 1ezphone.com. Anyone knows what happened with this site?. Thanks! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Regards Mattias -- Mattias Andersson -------------------------------- Storskiftesv?gen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38

please can anyone help me knowing if i can install Linux and Asterisk on HP servers _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008 -------------- next part -------------- An HTML attachment was

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

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

Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? Thanks. -- Rgds, -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: hansyin at gmail.com MSN: hansyin at hotmail.com Skype: hans_yin_vancouver