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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

...#39;t understand exactly if that is a problem of SAmba 3.x (As root, I can change the password) 3. Finally a question more: How would I do from WIndows if an user change his password and automagically change in Linux too? Must I compose a script for his connect with samba server? SErgio BELKin Obed Liberty Software Libre al desktop http://obed.com.ar ------------------------- Baje el manual para el nuevo usuario de GNU/Linux de http://www.obed.com.ar/doc/ ------------------------------------------------------------------

How To Installing tcng at slackware 10.1

HI I want to write my own SIP client (compatible with ASterisk) is there any good API available for this purpose . any help in this regard would be very helpful 4 me Obee _________________________________________________________________ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail

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, 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

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:

...is no complain from the caller). I have set following values in zapata.conf echocancellation=yes (also tried different powers of 2) echocancelwhenbridged=yes is there any other setting or not ??or this is a Softfone prob.??(i m using SJphone ) any help in this regard would b very helpful for me Obee _________________________________________________________________ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail

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

...nt address rejected: unverified address: host ns.mahan.org[private/dovecot-lmtp] said: 550 5.1.1 <mahan at mahan.org> User doesn't exist: mahan at mahan.org (in reply to RCPT TO command); from=< pmahan at silver-peak.com> to=<mahan at mahan.org> proto=ESMTP helo=< NAM02-BL2-obe.outbound.protection.outlook.com> 'mahan' does exist on ns.mahan.org. So I am confused to why lmtp is failing to find this username. Thanks, Patrick -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://dovecot.org/pipermail/dovecot/attachments/20...

...t 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 -- -------------------------------------------------------------------- Alejandro Cabrera Obed Interconexion SINTyS Sistema de Identificaci?n Nacional Tributario y Social Consejo Nacional de Coordinaci?n de Pol?ticas Sociales Presidencia de la Naci?n Julio A. Roca 782 - Piso 5 Ciudad Aut?noma de Bs. As. Tel: (54 11) 4343-0181/89 interno 5172 4334-3676 4342-5648 acabrera@sintys.gov.ar NOTA...

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 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

...it runs into this error: setup is unable to open the data file X:\~MSSETUP.t\~msstfof.t\Word97.stf';run setup again from were you ariginally ran it im stumped, what should i be doing here differently...thanks ............................... Daniel W. Schar Dept of Biology (OBEE), UCLA 621 Charles Young Dr. South Los Angeles, CA 90095-1606

...B : /etc/shorewall/interfaces : net eth1 detect tcpflags,dhcp,routefilter,norfc1918 loc eth0 detect tcpflags vpn br0 /etc/shorewall/zone : net Net Internet loc Local Local Networks vpn Vpn prova vpn obe /etc/shorewall/masq : eth1 192.168.10.0/24 /etc/shorewall/policy : loc net ACCEPT # If you want open access to the Internet from your Firewall # remove the comment from the following line. fw net ACCEPT net all...

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

...n > accomplish a particular goal (drop communication between > virtual-networks or allow them): Sure, that's simple if you're going to start/stop all virtual networks together as a group. It's more complicated if you want each network to operate independently of the other (i.e. t obe able to start/stop each network without affecting the others). Possibly the way to do that would be to create separate chains for the allow and block. You're welcome to write a patch for it :-) > > (Notice that I did not insert or delete any rule; just changed the order): > > - All...

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: