similar to: 12 CANCEL's followed by 12 INVITE's in 5 secs

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients. In a proper context, I have mentioned extensions 107 as simputer@bogus.com Asterisk Server-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on asterisk terminal ---------------------

Greetings, Today, there was something on the news of the 'simputer'; a very 'simple computer' that currently is being developed in India. (http://www.simputer.org/). (palmtop computer priced some 1000 Rs -which should be roughly some 160 euro-. It's based in linux and one of the ways it tries to be as cheap as possible is by avoiding all kind of licensed technology. Now, when

Hi list, i have a problem mounting shares on a single client with diffrent usernames. if any user on the client gives a wrong password all existing cifs mounts are closed by the server. Since this problem occured in a productive environment i set up a testing env and tried to keep it as easy as can be: Servername: serverA Servicename1: home1 Servicename2: home2 Clientname: clientA Username1:

Hi, i have sporadic connection problems with my rsync server. I have _no_ firewall involved and there is _no_ max connections set. rsync on the clients is started from cron via a bash script. There's also no permission problem. rsyncd runs as daemon (is not started from inetd). It sometimes works, but every third or fourth rsync fails with this message: rsync: failed to connect to

Dear all Recently, i did some work to test small files performance for gnfsv3 transport. Following is my scenario. #####environment##### ==2 cluster nodes(nodeA/nodeB)== each is equipped with E5-2650*2, 128G memory and 10GB*2 netcard nodeA: 10.254.3.77 10.128.3.77 nodeB: 10.254.3.78 10.128.3.78 ==2 stress nodes(clientA/clientB)== each is equipped with E5-2650*2, 128G memory and 10GB*2

Hi We are provisioning some default features to our customers like "automated answer when outside labor time", "rerouting when Subscriber Absent", and so. These are macro calls embedded in key points inside the dialplan. Since not all customers need/want all features and they (the features) are customized, I named them like [macro-feature1-ClientA], [macro-feature4-ClientF],

this is the case: serverA [ CentOs 5.6 kernel 2.6.18-238.12.1.el5.centos.plus, Samba ver. 3.5.21 ] serverB [ CentOS 5.6 kernel 2.6.18-348.1.1.el5.centos.plus, Samba ver. 3.6.6-0.129.el5 ] clientA [ WindowsXP ] clientB [ WindowsXP ] The serverA shares via Samba the resource [test] [global] workgroup = DMIT netbios name = SAMBA server string = DMIT domain server interfaces = eth0

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hi Sorry if this comes twice; i sendt first version from non-member address. I'm learning use Asterisk but cannot solve following problem: i have Asterisk v1.0.7 (DEbian) and Linphonec v1.2 (Debian). Every time i try to register within LAN i got 'Forbidden' message from Linphonec. Where to start searching for reason for this failure, is there more debuggin options

nautiluz wrote: >I am starting project to develop open implementation of some (naturally open >codec) for simply designed embedded devices which can be used by small to >big VoIP operators or hobbyists which wants to build their own small and >low cost VoIP phone. >Speex seems to be great choice. And greater will be if there should be >some guys who want to help ;-). Sorry

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

Hi, I''m looking to run several installation of a rails app. Each one has certain unique elements, like database.yml some and some css stuff, possible some templates as well. I''ve set this up like /w/sites/clienta /w/sites/clientb /w/sites/clientc Inside these directories there will be mostly symlinks to: /apps/myapp Urls for these apps will be: http://clienta.example.com

Hi, I have some problems and doubts connecting two asterisk servers. I have one asterisk (serverA), with 1 sip client registered (clientA). I have another asterisk (sever B), with another client (clientB). Now I want to call from client A to B and from B to A. Searching in google i find many configuration examples. For instance:

Hi Muhammad From: "Muhammad Talha" <talha@worldcall.net.pk> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, December 22, 2004 2:07 PM Subject: Re: [Asterisk-Users] Zaphfc/BRI Configuration help > Thanks for lan for your reply can you share your extention.conf > setting . > >

i'm planning to migrate a call center to asterisk, i don't understand if i can launch a resident application on the agent's client in relation with the queue the agent's is answering. For example: I have - queue A - queue B - queue C Agent 100 (logged in A.B,C) Agent 101 (logged in C) When Agent 100 receives a call from the queue A i'd like to launch his browser and point

Hi Guus, Am Freitag, den 25.09.2015, 17:04 +0200 schrieb Guus Sliepen: > Ok, that means by default the UDP NAT timeout on the Cisco is extremely > short. > > > I check the manual of the the Cisco NAT for any TCP/UDP > > timeout settings, but there is no way to modify anything like "keeps > > TCP/UDP connections alive". > > It wouldn't be called

Is there a linux sip or iax phone that will autoanswer and connect to the console or soundcard? I found linphonec but it does not autoanswer from what I can tell. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050916/8bc76bbb/attachment.htm

Hi, I've got this PXA270 board set-up with Linphone 1.2.0 and am trying to get linphonec to work with Asterisk. I have the echo test working, but when I dial in to this, to voicemail or anything else using Playback() to play a sample, I hear nothing for ages (10-15 secs) and then little sections. With the echo test, I get the tail of the message (...pressing the pound

Hello, I am trying to build a Samba standalone server to serve file sharing in our office for both Ubuntu and Windows clients. Ubuntu user will use mount.cifs to mount the share. Windows (XP/7) will use workgroup to access the share. I am using Samba 3.6.9 on CentOS 6.3 (64bit). Now I am having problem to setup correct ACL so please help me. Here is my situation: I'll have directories