search for: jezek

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk...

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho,

Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold

Hi, have you some idea, how to make "roaming line" with Asterisk? i.e. is possible to have phone line assigned to user if migrating from one office to another? thanks PJ

Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is "caller id name" transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ

Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its through the Internet. I want to be sure no one makes calls as being me, and that my calls aren't intercepted. Is it possible to define encrypted trunks? And should I define the trunk in SIP, IAX or something else? Thanks Joao Pereira

A week ago, I tried give realtime priority to asterisk proces using -p switch, asterisk was running inside astcanary, but yestarday asterisk probably starts eating all cpu and lock any access to computer, only ping was possible, so, anybody have experience, that ascanary process does really work to lower process priority in case of overloading? PJ

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

Hi all!! I have downloaded the asterisk from svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the asterisk 1.4 subversion). I also downloaded the patch for cellphone and make it work fine. Then I bought the tdm11b board to have phone connection in my computer. I installed the hardware for zapte and the libpri modules in my Mandriva 2007 and the lights of the pci card

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

...ndova tesarkova hovorka selementova vodickova hrivna kova lezak teprt rosulek krumpl niesytova chadim [inzerce] comment = Inzerce path = /public/inzerce browsable = yes read only = no create mode = 0664 create mask = 0664 directory mask = 0775 valid users = hajkova jezek chadim kotherova write list = hajkova jezek chadim kotherova [final] comment = Material pripraveny k tisku path = /blackhole/final browsable = yes read only = no create mode = 0664 create mask = 0664 directory mask = 0775 valid users = chadim niesytova krumpl...

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

...where to look for solving this issue? When we compare this log with the log using ntlm_auth we don't see the entered username inside the log, just the apache user! Why is that? Greetings from Switzerland [1] http://adldap.sourceforge.net/wiki/doku.php?id=seamless_authentication -- Patrick Jezek | Leimeneggstrasse 25 | 8400 Winterthur VoIP +41 52 508 24 34 | Mobile +41 79 270 22 68 http://cms.jezek.ch/blog | patrick at jezek.ch | GPG 0x883AF385 Hilf Frank im Ozean zu ?berleben: http://daddelbox.com/2/ftf Mach einige Fliegen gl?cklich: http://daddelbox.com/2/hf -------------- next part ----...

Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so

...manager in asterisk) thanks PJ * dialplan: '953' => 1. NoOp(${CALLERID}) [pbx_config] 2. Dial(SIP/953) [pbx_config] asterisk log: -- Executing NoOp("SIP/324-655d", ""Pavel Jezek" <324>") in new stack -- Executing Dial("SIP/324-655d", "SIP/953") in new stack -- Called 953 -- SIP/953-cd50 is ringing YAACID config: channel: SIP/953 advanced event: newchannel or dial , channel type: channel or destination...

Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ

Can anybody tell me how I can set-up multiple access points with overlapping coverage, so that a moving WiFi phone user can continuesly use the phone. bye Ronald Wiplinger