similar to: help on asterisk sipp

I am trying call transfer with asterisk. blind transfer (#) is working perfectly, but attended transfer doesn't fonction (*2). I don't know what is the problem. Anyone could help? _________________________________________________________________ Lancez des recherches en toute s?curit? depuis n'importe quelle page Web. T?l?chargez GRATUITEMENT Windows Live Toolbar aujourd'hui !

I am trying to test the call parking, but It doesn't fonction :(these are my config files.extensions.conf:include=>parkedcallsexten => 4000,1,Dial(SIP/4000,60,tT)exten => 4001,1,Dial(SIP/4001,60,tT)exten => 4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext => 700 parkpos => 701-720 context => parkedcalls [featuremap]blindxfer => # disconnect => *0

Bonjour, En utilisant le logiciel Samba je me suis rendue compte qu'il est possible de supprimer n'importe quel dossier ou fichier, qu'il soit ou non prot?g? contre l'?criture, ce qui pose un ?norme probl?me de s?curit?. Avez-vous connaissance de ce probl?me ? S'agit-t'il d'un simple probl?me de param?trage ? Dans le cas d'un bug, avez-vous une solution ? nous

I am using Aserisk as a SIP server to interconnect differents PBX in differents sites. I am now looking for a tool that can test the performance of this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone

bounjour je dispose de differents commutateurs de LG (IP LDK) sur differents sites. je voudrais savoir comment je pourrais interconnecter ces differents IP LDK a un serveur Asterisk via IP ( ceci sous entend que chacun de ces commutateurs dispose d?j? d'une carte VOIBE). Mecri d'avance pour l'aide _________________________________________________________________ MSN Messenger :

Hi everybody, I try this : [incoming_from_fxo_card] exten => s,1,Answer() exten => s,2,Background(filename) exten => s,3,Dial($(INTERNAL_SIP_TEL)) But * wait the file is finish before make Dial to SIP channel. Background(filename) (from voip-info.org) => Starts playing a given sound file, but immediately returns, permitting the sound file to play in the background while the next

hi, I can''t get a freeswan 2.02 ipsec x509 connection at work can somebody help me? ************************************************************************************* global situation ************************************************************************************* the linux gateway (chivas) is a single machine 192.168.1.250 with a local net 192.168.1.0/24, a dyn IP via a DSL

Hello list, I have a problem with my samba shares. I have a server with samba installed on it (3.0.7-Debian). I have workstations under wxp and workstations under linux. I have a common share which looks like this : [Archive] available = yes valid users = user1, user2 comment = Repertoire Archive browseable = yes write list = user1, user2 writable = yes admin

English version : Hi i've some problems with samba 2.5 and MDK 8.2 each time I log on a WIN2K SP2 or SP3 client I get the message : your password expires today would you like to change it ? yes no Even if I change it , I get the message one more time when I log on the client the next time.. I read the o'reilly and the how to and I didn't see any param for password expiration in

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The

Hello, I'm stressing an Asterisk 11 platform with incoming calls from sipp 3.1. I've dedicated a context to sipp in my dialplan. Everything works OK expect that calls from sipp comes in with a CallerID set to sipp and this sipp value is stored in CDR. 1. I can change the value of the CallerID but how can I have the calls from sipp traced in CDR with a customized src field value ?

Hello everyone, Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()

You maybe have to create a SIP user called like it is declared in your UAC/UAS xml file. I think it should be 'sipp' or something like that... -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: <asterisk-users@lists.digium.com> De: "C. Johnson" <javadude@cedrick.net> Envoy? par: asterisk-users-admin@lists.digium.com Date: 31-05-2004 08:03 Objet: RE:

Hi, I want to put on the wiki a tutorial I wrote about Cacti on CentOS4 My UserName is UgoBellavance Regards, -- Ugo Bellavance (ugob at lubik.ca) Consultant en S?curit? Informatique Lubik Inc. Site Web: http://www.lubik.ca # T?l.: 514-907-3253 # Fax.: 1-866-334-1426 Telephone IP (SIP): ugo at sip.lubik.ca Protection de courriel par LastSpam (www.lastspam.com)

Hello, Using Dovecot's deliver MDA, all mails are stored with the line "From <sender> <date>" appended. When forwarding these mails as attachements, the recipients get the source code. When Postfix delivers emails directly in the Maildirs, this first line is not appended, and there is no problem when forwarding the mails. Why is this line appended by deliver ? Is

Hi! R users I would like to ask you where could we find the VGAM Package. I don't find it in the list of packages. Thak you for your help Lassana KOITA Etudes de S?curit? et d'Exploitation a?roportuaires / Safety Study & Statistical analysis Service Technique de l'Aviation Civile (STAC) / Civil Aviation Technical Department Direction G?n?rale de l'Aviation Civile (DGAC) /

Hi all, I'm trying to use SIPP (http://sipp.sourceforge.net/) to stress-test our asterisk installation. We have a very simple dialplan that uses FastAgi. I'm finding that all calls to "GET VARIABLE" from the FastAgi are returning null when the dialplan is invoked from sipp -- and they work fine when invoked from a softphone on the same machine, for example. Does anyone have

Hi there! Does anyone knows how to test Asterisk load with sipp? I am using uac.xml to call a 'playback extensions' via a SIP channel. When I increase the Call rate (about 20cps), I begin to have INVITE/200/BYE retransmissions meanwhile the RedHat box is not loaded at all (made a TOP). Where is the pb? [root@10.54.196.38 sipp]# sipp 10.54.196.32 -s 9001 -sf uac.xml -d 100 -i 10.54.196.38

Hey I'm looking for an advanced scenario for sipp, that can be used for testing asterisk. Mainly I'm interested in making random calls between sipp pseudo-users. Did anyone try to do something like this? Or has anyone got an example scenario with working loops? Thanks