similar to: Réf.: Call generator

Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in

Hello, Does someone know how to setup snmp monitoring on asterisk. I?ve plan to deploy 50 asterisk, so I need some monitoring tools. I try with nagios as I read in the wiki, there is some project on it, but I can?t reach the end. Can someone help me? Thanks. GIBERT Fr?d?ric Ste VigiNetworks Mobile: +33 6 72 08 35 16 -------------- next part -------------- An HTML attachment was scrubbed...

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:

Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

I tried that but it still doesn't work... I think I don't have the correct approach. Have I to install any ISDN drivers (=modules ?) BEFORE dealing with CAPI ? If yes, why shouldn't I use the hisax drivers (which are kernel ones) instead of fcpci drivers (which doesn't seems to work, by the way...) And finally, how is it possible to link the two modules together? As you can see,

Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance.

The origine of my message is a problem with my local LDAP server. last thursday I upgraded my RH 8 with the glibc update from RedHAt, after sometimes the LDAP server is unreachable. In log : Mar 2 11:40:02 coradm01 slapd[5342]: warning: cannot open /etc/hosts.allow: Too many open files Mar 2 11:40:02 coradm01 slapd[5342]: warning: cannot open /etc/hosts.deny: Too many open files Mar 2 11:40:02

OK! Thank you for your answer. Merci de ta r?ponse... RB g.patel@wanadoo.fr.invalid (gerard patel) on 17/10/2001 16:18:46 Veuillez r?pondre ? wine-users@winehq.com Pour : wine-users@winehq.com cc : (ccc : Roland Baudin/ALCATEL-SPACE) Objet : Re: R?f. : Re: Dead keys on Wine On Wed, 17 Oct 2001 12:36:46 +0200, Roland.Baudin@space.alcatel.fr wrote: >Ok, but isn't there any

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><div>Finally, i will get back to a RedHat 9 distrib as I see that it works with that distribution...</div></FONT>

If the first LDAP server faild, the second can be used directly. This server is a PDC server with more than 100 people connected and some application required domain authentification for running. For me is a critical server. ----------------------------------- St?phane PURNELLE stephane.purnelle@corman.be Service Informatique Corman S.A. Tel : 00 32

Mateus, here is a more complex (working !) example : vfs objects = recycle recycle:name = .recycle ; max-size (in bytes) of files allowed in the recycle bin recycle:maxsize = 2000000 ; keep directory trees ? recycle_keeptree = True ; files to exclude from the bin recycle:exclude = *.tmp *.temp *.swp ; root dirs to exclude from the bin recycle:exclude_dir = tmp ; include file versionning in the

Hi everybody, got it from svn: dtmf_2833_1.pcap */asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN *>*

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

damn.... now everything works samba recognises user "test_user" in group "users" AND "kids".... i dunno why ?!?!!? i did nothing, i just removed "valid users" from this share and reloaded smb-conf...nothing special ! if i could reproduce it, it would be better then seeing it working now and not knowing why.... but thx very much for your patiance greez

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 ?

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

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

Hi, I just installed Sipp 3.3?on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. SIpp output: ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- ? Start Time???????????? | 2013-05-20?22:53:08:637?1369083188.637273??????????? ? Last Reset

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()