similar to: Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID

Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by

Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send

hello, the last 3 times I installed C5, the MBR was unchanged and nothing was written into /boot/grub except splash.xbm.gz there were no stage* files, nor a menu.lst I know how to fix that. Would I have better luck using a partition mounted as /boot? Anyone succeded with Grub that way? I prefer not having to do surgery to get C5 to boot :) -- Mark New Packages for C5 ---------

Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? Cheers, Igor Neves.

Hello, I have 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my <mailto:*@home> *@home 2.8 running on top of CentOS. Both FXO Ports are on ONE Digium card. What I would like is: If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4 If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234 on my *

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and

Hi, Any one know why the glibc-profile is obsoleted on centos 5 (glibc 2.5)? I've checked the glibc.spec file and find that the --disable-profile option is enabled in the configuration command. So how the original profiling functions are implemented on glibc 2.5? Is there a new way to do the same/similar thing? Thanks a lot.

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

Hi List; If I configured one SIP account or one IAX account [sipuser1] or [iaxuser1] then how many calls can be originate/terminate using the same account [sipuser1] or [iaxuser1]? In other words, can 10 IP Phones (users) do a calls via Asterisk using the same account (SIP or IAX2)? If yes, how can I control the number of calls per account? Regards Bilal

>Wine Is Not (an) Emulator. yup i know that :) >This means that it cannot run windows - it /is/ >windows as far as your >end-user applications are concerned. no issues with that >This means that you should not use your windows boot >as a 'fake >windows', but instead install applications properly >and use them. i usually run a setup of all my windows app in wine

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Hi all, > > I've got two problems compiling the current CVS FLAC sources on OSX. > > Firstly, the configure script can't find the OGG libraries which were > installed from MacPorts. I have tried: > > ./configure --with-ogg-includes=/opt/local/include \ >

Josh Coalson wrote: > > The test file is here: > > > > http://www.mega-nerd.com/tmp/flac_char.ogg > > yep, this is definitely a bug, thanks for the test case. requires > a tiny fix to the ogg seek algorithm which I will check in tonight. Oh cool. I was wondering what happened to that issue. Erik --

hi FLAC list! Does anybody know where I can request a new feature (if that's possible of course) for the FLAC frontend, included in the windows installer? The feature I would like (if the developers agree and want to implement it) is the possibility to add FLAC files to the list in the frontend and re-encode them to FLAC. This is useful if you want to re-encode some files by using the

Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into

Hello - I running Samba as a PDC on FC6 with roaming profiles. I need to setup a custom Windows logon/welcome message... to tell users want they can expect using this domain. Is it also possible to place different PDF files on the users desktop when he or she logs on, but only referencing one source file, so I don't have a copy for each user? What is the best approach? Thank You. Ralf

Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf.

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

Hi to all i think this is not a new problem for this forum....but its newest for me as i m a new linux lerner. Even if it is new plzz....reply me ur answer..n if its already asked n have solution..plzz forward the solution. My problem is here mentioned: I have fedora core 4 as a linux server. there r two external links connected to this. the settings are as: eth0 ->for internal (that is

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext