similar to: Restricting amount of incoming calls

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled "Unnumbered Priorities." http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip -MC

Hi, does anyone know the prices for g729 codecs from Digium? I sent an email a while ago to them but haven't got any response so far. Prices are per unit or volume? Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, We are considering a transition from OpenVZ to Xen for our web server infrastructure. The primary task performed by the virtual servers is to run a number of Drupal sites with a MySQL backend. The webserver and MySQL servers are on separate virtual hosts. We also have a number of smaller hosts running on the same hardware (a couple of dev-servers, a logging-and-monitoring server, a secondary

Hello, Thought people might be interested in this. I want my voicemails emailed to a person and not stored on my asterisk server. However, I want them to have a sequential number. I found that if I set the option delete=1 in my voicemail.conf file for the mailbox, then the numbering would keep being restarted. I wrote this shell script to fool Asterisk into numbering my voicemails sequentially

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

Hi! Im looking for a very basic example for the following simple problem. I've been searching voip-info.org and looked in the ORA book without a clue. I have a SIP account at sip.provider.com and my own asterisk server. What I want is the following: I. Register my phone to my asterisk server, not directly to provider.com II. My asterisk server should ring my phone when somebody calls me

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

List, Does anyone know how to add the dst Country to the CDR's via Macro (preferably). For example, I will add a column in the cdr DB table and when someone dials 01158212XXX. I want the CDR's to show Caracas as the destination in this new column. I have all of the International destinations in my extensions.conf like the example below: [macro-dialout-intl] exten =>

Currently, our organization grants some employees remote access to SAS and other software through a secure Citrix-based network server. We would like to install R on a dedicated network server and enable remote access through Citrix. Is this possible? If so, what additional software would we need? Any advice would be greatly appreciated. Erick Edwards

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition?

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

I have no idea as to how to size an email server. I was approached by a customer that wanted a single server with RAID 1 disks to handle about 5 million emails a day. In general terms, what parameters should I take into account to size the hardware specs when the average email is about 10kb, the smalles email is 2kb and the largest email is about 5meg (with attachment) thanks, --

Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, We have a page on the wiki just for phone reviews, but I think it needs a bit of format change. Instead of individual reviews for each phone, I think each person should review all phones they have worked with and list the phones they have had access to and rank them in relation to each other. Also each review should have a date so the reader can see how fresh the data is to current.

Hi, today a customer asked how to use asterisk with 12 to 16 FXO ports. I can use a channel bank with 16 FXO ports and connect the channel bank with a T1 cable to a T1 card in the Asterisk Server. Asterisk will then send the calls to the Voip provider over the internet. However a 16 fxo port channel bank is about USD 1500 + a t1 card USD 500 + a USD 1000 computer = 3 thousand us dollars + my

Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, One customer has Centos 5 installed. He wants to have a web based frontend to the vacation program because he is in charge to enable the vacation msg for all the users in leave. I tried webmin but the webmin vacation module points to a nonexistant link. And the usermin module is very old and requires the user to do it by themselves. Suggestions? --