search for: openpbx

at http://wiki.openpbx.org/tiki-index.php?page=Easy+route+to+building+OpenPBX.org there is the following adduser command: adduser --no-create-home --ingroup openpbx --disabled-password --disabled-login openpbx This does NOT seem to be the right format for Centos. So far, using man, I have come up with; adduser -M...

I am interested in getting Openpbx up on Centos. The install information is at http://wiki.openpbx.org/tiki-index.php?page=Getting+Started And it is quite a bit beyond my current skills, so I am looking for some one that has done it and can help step me through it. I want Openpbx over AsteriskNow or Trixbox to get T.38 support.

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

at: http://wiki.openpbx.org/tiki-index.php?page=Easy%20route%20to%20building%20OpenPBX.org A number of dependencies are listed, with a Debian slant. I have found most of these in our standard Centos repos ( snarfed spanDSP from trixbox repo). But the following I have not figured out: * linux headers for your kernel...

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has improved since the fork but they still have the same complaints: the Asterisk core is a &quo...

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

Hey, i'm trying to patch asterisk to work with ncs. I get this error: root@fox:/Kit/openpbx/asterisk-1.0.0# patch -p1 <../conf_patches/ncs_dtmf_combined_v03.diff can't find file to patch at input line 3 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -------------------------- |--- channels/chan_mgcp.c 16 Jul 2004 21:22:55 -0000 1.60 |...

Hey, i'm trying to patch asterisk to work with ncs. I get this error: root@fox:/Kit/openpbx/asterisk-1.0.0# patch -p1 <../conf_patches/ncs_dtmf_combined_v03.diff can't find file to patch at input line 3 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -------------------------- |--- channels/chan_mgcp.c 16 Jul 2004 21:22:55 -0000 1.60 |...

Please note that we are available to fix the current REMOTE crash that affects Asterisk/openpbx/trixbox and crashes these systems via a malformed packet please contacts use if you need a hand to patch your systems. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part -------------- An HTML attachment was scrubbed......

...hannel stuff. My guess is to go with the lower 100 or 250 Hz option but that is only a guess. The 1KHz sounds like it will conflict with the Zap 1khz timer (or am I wrong about that). Does anyone know what the prefered settings are for Trixbox or AsteriskNOW (or the asterisk code forks e.g. OpenPBX)? Please let me know what your experience has been. Aloha, Mark C IS Director - Payroll Services Hawaii, Inc. http://www.psh-inc.com FWD: 293625

Hi, I am using Asterisk for a call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ"): The main points listed on Asterisk's "CONS" that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT traversal; * Asterisk doesn't use SpanDSP; * Use o...

...days away! Meeting Start: 09/08/2007 - 11:30am Hello all Twin Cities Asterisk Users, It's time once again to have another meeting. I've not had much time to prepare, but I'd really like to review and install with the group at our next meeting the software package formerly known as OpenPBX and now known as CallWeaver. This release now claims to have full t.38 support for faxing over IP. This has got to be one of the biggest issue facing professional installations at many business locations. OpenPBX and now CallWeaver are software forks of Asterisk. Lets think of the meeting as vi...

Is an RPM for Asterisk (the PBX system) available for CentOS 5? It looks like RPMforge is supposed to have one, as I can see dependent packages like asterisk-sounds, but the base package seems to be absent from the repository.

Does anybody know if Avaya has a test SIP firmware available for 4620 and 4640 IP phones? The 46xx SIP image from their website is a combo download with SIP for the 4602, and h323 for the the 4620 and 4640. It looks like they demo'd a SIP image for the 4640 as far back as 2004: http://www.sip.org/von/2004/boston/slides/DSC_0042.php Thanks, Andy -------------- next part -------------- An

Hi, I thought I remember seeing somewhere on the Asterisk website a card that had 16 ports fxo or fxs, that was user selectable with straps on the card. Am I going crazy, I can't seem to find it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/86c8ae69/attachment.htm

Hello List! I wanted to share to everyone the following compatible connectivity products that my company installed in our Asterisk based soft switch. I already sent these to the Asterisk.org site many days ago but for some reason they still have to post it. 1. Sangoma A101 single port E1/T1/PRI Card 2. Sangoma A102 dual port E1/T1/PRI Card 3. Sangoma A104 quad port E1/T1/PRI Card 4. Sangoma

...--------------------------------------------------------------- Since the configuration method is a bit too much for me, here's part of chan_capi "Makefile". I think I've been blind as I haven't found the documentation for WHAT needs to go WHERE in this Makefile... .PHONY: openpbx INSTALL_PREFIX=/usr/lib/asterisk ASTERISK_HEADER_DIR=/usr/src/asterisk-1.4.0/include MODULES_DIR=/usr/lib/asterisk/modules CONFIG_DIR=/etc/asterisk //---------------------------------------------------------------------------------- If anyone has any idea what I'm doing wrong, please hel...

Is it possible to show CallerID names for dialplan applications? When I call from phone-to-phone, it shows the CallerID from sip.conf or iax.conf, but I don't know of any way to show CallerID Name when I call the extension for an application (voicemail for example): exten => 1000,1,Answer exten => 1000,n,VoicemailMain I'd like the display to read "VOICE MAIL" when I

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08