similar to: Problem with valid users in Samba

Hello, I have been using Fedora Core, Samba, and Active Directory to provide authentication services for Windows based users for a few years now, but as an experiment I wanted to accomplish the same service with SUSE 9.3 . I have been able to get this configuration to run successfully with RH9, FC1, FC2, FC3, and FC4 (buggy but works), but with SUSE I have stalled a bit. I feel I have

Hi, I am using Samba version 3.051 in an Active Directory setting with Windows 2000 server. Everything is working rather well with regards to file-sharing and authentication. However, the one thing that I noticed that I haven't been able to fix quickly with SWAT is the prevention of browsing the Linux file-system with users such as 'nobody' or 'bin'. For example... I have a

Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is 8888, how can fill the Dial field from the custom extension ??? Because if I put just 8888 or Local/8888, I don't succeed. Thanks a lot

Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card ???? Thanks a lot Alejandro

Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Regards Alejandro

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hi, I was using Samba 3.0.9 on Fedora Core 2 and decided to upgrade to 3.0.10. So I upgrade to Core 3 and installed Samba 3.0.10 and thought I could just copy my settings over to the new build and everything would run smoothly. I thought wrong. Everything seems fine until I enable Obey Pam Restrictions. If enabled I get a login error from XP stating: " Windows cannot locate your

People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with "apt-get" the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards,

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got

Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in voicemail sounds files (I have Spanish sounds). But now I have a problem because I have to use G.729 mandatory at peers, and I have GSM in voicemail sound files. I can't let Asterisk do trascoding because I have no a DSP in the CPU, and I don't want to degrade the PBX performance with trascoding tasks. So how can I

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with "envelope=yes" and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full:

Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

Dear, when I dial *30 in order to get instructions to blacklist an extension, Idon't get the menu but I get a new dial tone. What happen please ??? What can I do to solve this ??? Thanks a lot, Alejandro

Dear all, I'll implement a VoIP system using Asterisk + SIP with softphones; I need to connect LAN and VPN users (about 100-150). What version/installation of asterisk do you recommend tyo me ??? Does Asterisk@Home or Trixbox match to my scenario ???? By the way, I use Debian Etch as OS server. Really thanks. Alejandro --

Dear all, I use Trixbox as my PBX. Until a couple of days I've installed a GSM Gateway to communicate with our three cellular phones: 15 64227777 15 64228888 15 64229999 The GSM Gateway has just one SIM. I use the Free PBX web interface in order to set up the route and trunk parameters: Trunk: ******* Name: SIM1 Peer details: host=10.10.1.2 (IP from GSM Gateway) port=5060 type=peer

Dear all, I've read that Asterisk supports only the G.729 A audio codec. I have several Grandstream IP phones with G.729 A/B codec implementation. Does G.729 A/B mean both version A and version B, or A/B is a new version different from A and B and it's not supported by Asterisk ??? Thanks a lot Alejandro