search for: mandl

Hello List, I''m new to QoS/tcng/HTB and friends, so please forgive me if my question might be silly... After having read lots of HowTo documents I''m totally confused... The Challenge: ============== I''ll have to deploy several "mirror" download servers (Linux) which must be able to handle a huge number of HTTP download requests (about 10k to 20k unicast

...best regards Steve Smith -- F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Sysadmin/VOIP // sst at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------

...__________________________________________ Michael Keilhacker, MBA, M.A. Logistics and Supply Chain Management Technische Universit?t M?nchen ? TUM School of Management Arcisstra?e 21 ? 80333 M?nchen ? +49-163-5454918<tel:%2B49-89-289-28203> ? michael.keilhacker at tum.de<mailto:christian.mandl at tum.de> www.log.wi.tum.de<http://www.log.wi.tum.de> [[alternative HTML version deleted]]

Hi all, This issue is giving me a lot of grief with my customers. I have 5 asterisk servers running in production, each one with almost 70 simultaneous calls at peak hour. Most of my customers complain that their calls drop after 20 seconds or so. After running through my cdr's, I see that the number of 20 second calls is MUCH larger than any other number. (see below) billsec count(*) 1 924

Hello, Despite of reading a lot of descriptions I don''t understand the network configuration. I simply have too little experience. I work remote with the machine via ssh. My dom0 is running. I copied an existing installation into a LV and created the following file: # cat /etc/xen/domu-sarge # -*- mode: python; -*- kernel = "/boot/vmlinuz-2.6.18-4-xen-686" #kernel =

Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the

...best regards steve smith -- F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Sysadmin/VOIP // voip at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------

Hello all, I need that Asterisk Always use Reason in a CANCEL. How to do? thank you *Fran?ois * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100521/f3a91f36/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: francois.vcf Type: text/x-vcard Size: 400

Hi, I'm currently programming an interface for my Asterisk service. I've noticed an issue if someone sets up call forwarding on their sip phone. Asterisk receives a 302 "Moved Temporarily" message, and forwards the call successfully. However, the CDR is not correct. If I set up call forwarding to a mobile on extension 201, and then place a call from extension 202, the call

...ork as well. When packet comes off from wire, I mark it with 3 at mangle PREROUTING. Since it is a ping to 192.168.8.88, it should be a local process. Then the ping is successful. But from my testing, no. Another possiblity is packet is route to test2 routing table after mangle OUTPUT and before mandle POSTROUTING. I am getting confuse :) Please advice. Thank you Kaiwen

Hi, How can I match an extension "ending with 3" (just an example but applicable to any other digit, including * or #)? exten => _ZX.3,n,... exten => _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: "$")? Thanks, Vieri

Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid:

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a

Hello everybody i am Ashish here. i am new to this mailing list. so dont know rules and regulation, just trying to post my problem of voicemail.conf Actuallt right now i am using Asterisk 1.2 on my LAN environment. i am able to call all my extension very nicely. Right now i am trying to deploying voicemail facility for all extensions, so if anybody is not present, then he/she can leave message,

Hi, Does anyone have a suggestion how I can analyze the concurrent usage of ISDN channels? A client complains about their clients sometimes getting a busy tone when trying to call them. I suspect they just need to add an additional ISDN2 line but I need some conclusive information that they are indeed maxing out their ISDN channels. Thanks, Patrick

An attacker is scanning my Asterisk Switch to gain illegitimate access to VoIP call functionality. Using a sip scanning tool, *it* sends REGISTERs with random identities. And when it discovers one identity subscribed in my switch, it tries to authenticate with random passwords using this user name. For the moment, I have replaced this account. And also blocked the IP it has used but each time

Working with a new client that has a ton of these phones, and in a new installation the phone is registered, can place and receive calls with no issues, but has a "locked" picture of a phone in the upper right corner. Any Linksys experts know what this means? I have searched the admin guide and googled to no results... really just an annoyance I suppose, but I would like to know

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

Greetings list, We recently installed some Linksys SPA962 + SPA932 sidecars into a client's offices. The BLF functionality works fine, but call pickup using the BLF is something of an issue. Following the advice on voip-info.org, I configured part of their dialplan as follows: exten => _**2XX,1,Pickup(SIP/${EXTEN:2}) exten => _**2XX,n,Dial(SIP/${EXTEN:2},15,tw) exten =>

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;