similar to: Centos kernel 34 vs. 42?

Hi list! I'm totally fed up with bristuff (or it's instability with a simple HFC-S card), 2 out of 3 times when people try to call they get the information tone that the number is not connected. I would like to try vzaphfc and I am looking for information on it. >From previous posts I found that the only place where the sources seem to be maintained and available is at the debian

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure

After upgrading from 1.2.25 to 1.2.26 i noticed that IAX -> IAX calls always result in Asterisk just exiting without any message. Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing 2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd call. Went back to 1.2.25 for the moment

Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them

Remco Barendse <asterisk@barendse.to> wrote: >> I'm not running trixbox but normal Centos 4 with asterisk installed. I tried to find some further info on this but couldn't find any. Do audio problems occur with normal Centos and the latest kernel version too? (In other words, should every centos user downgrade??) << I can't give you a definitive answer, but can

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Hi! I installed shorewall shorewall-2.1.7-1 on a newly installed box. When using 1.4 versions I could enable shorewall in ntsysv, now shorewall doesn''t show up in ntsysv? I am running shorewall on a RedHat Enterprise Linux rebuild (www.taolinux.org). Is this a bug or a feature?

I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

Hi list! Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk? Any experiences / caveats? If anyone would be willing to share the dump of their IP600 config file, i would really appreciate it. Is there anything special i should put in my asterisk config? Thanks !!! Remco

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

Could anybody help me with the correct rules to allow a client to connect to an NFS share on my Linux box/firewall? >From what I understand from some howto''s on the net is that there are several daemons that are required to make NFS work. Some rules I found in the NFS howto: iptables -A INPUT -f -j ACCEPT -s 192.168.0.45 iptables -A INPUT -s 192.168.0.45 -d 0/0 32765:32768 -p 6 -j

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm