Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when you go from g711 to g729 for example? Or when you go from SIP to IAx? 2- What is the best GUI tool to configure * ? 3- Do I need to install a PCI (fxo or fxs) to have meetme, music onnhold etc? 4- If I have a SIP device behind a firewall the supports SIP transformations (sonicwall pro230) and the * is outside the firewall, do I have to open ports 5060 anyway? What about the audio? Regards Fabian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050315/9c614253/attachment.htm
On Tue, 15 Mar 2005 11:56:18 -0500 "Fabian Borot" <fborot@hotmail.com> wrote:> Hello all > I have been learning * from almost 1 month now. It looks >really powerfull. I > have some problem trying to find previous post, or >solutions to common > problems, advice to newbies etc in this mailing list. >There is no a > forum-like tool to search thru the posts by keyworks for >example. Please > correct me if I am wrong. >Go to Google, in the search box type "site:lists.digium.com" without the quotes then type in what you want to search for. THis will limit all searching only to the Digium lists for asterisk.
This is what you need. Google allows you to enter a parameter called 'site:' when you do this it searchs that site only. The list is archived so you always have it available. Search at google with the following... site:lists.digium.com <some parameter> This will search the archive and you will get lots of results. Setup info and details of asterisk can be found at the Documentatio link on www.digium.com or on the Wiki at www.voip-info.org Thanks, Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Fabian Borot Sent: Tuesday, March 15, 2005 9:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Newbie Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when you go from g711 to g729 for example? Or when you go from SIP to IAx? 2- What is the best GUI tool to configure * ? 3- Do I need to install a PCI (fxo or fxs) to have meetme, music onnhold etc? 4- If I have a SIP device behind a firewall the supports SIP transformations (sonicwall pro230) and the * is outside the firewall, do I have to open ports 5060 anyway? What about the audio? Regards Fabian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050315/d4d186a1/attachment.htm
1 Transcoding is between codecs. ulaw to g.729 for example 2 I prefer AMP but unless you install it with *@home it could be a pain. 3. You need a clock source for meetme and other features to work so if you don't have any digium hardware you must use ztdummy 4. Unless you are using a VPN or STUN you must open the port in your firewall manually In addition if you search on google by 'list.digium.com : whatever subject' you could find the answers to a lot of questions. Have a good day, I hope this helps you get on your way. Henry ----- Original Message ----- From: Fabian Borot To: asterisk-users@lists.digium.com Sent: Tuesday, March 15, 2005 10:56 AM Subject: [Asterisk-Users] Asterisk Newbie Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when you go from g711 to g729 for example? Or when you go from SIP to IAx? 2- What is the best GUI tool to configure * ? 3- Do I need to install a PCI (fxo or fxs) to have meetme, music onnhold etc? 4- If I have a SIP device behind a firewall the supports SIP transformations (sonicwall pro230) and the * is outside the firewall, do I have to open ports 5060 anyway? What about the audio? Regards Fabian ------------------------------------------------------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050315/a750d094/attachment.htm
To search the list archives use this in Google: site:digium.com search-terms -----Original Message----- From: Fabian Borot [mailto:fborot@hotmail.com] Sent: Tuesday, March 15, 2005 10:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Newbie Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when you go from g711 to g729 for example? Or when you go from SIP to IAx? 2- What is the best GUI tool to configure * ? 3- Do I need to install a PCI (fxo or fxs) to have meetme, music onnhold etc? 4- If I have a SIP device behind a firewall the supports SIP transformations (sonicwall pro230) and the * is outside the firewall, do I have to open ports 5060 anyway? What about the audio? Regards Fabian
Or if google is too complex, http://asterisk.keystreams.com Roman Volf Keystreams Internet Solutions volfman@keystreams.com Robert Webb wrote:> > On Tue, 15 Mar 2005 11:56:18 -0500 > "Fabian Borot" <fborot@hotmail.com> wrote: > >> Hello all >> I have been learning * from almost 1 month now. It looks really >> powerfull. I >> have some problem trying to find previous post, or solutions to common >> problems, advice to newbies etc in this mailing list. There is no a >> forum-like tool to search thru the posts by keyworks for example. Please >> correct me if I am wrong. >> > > > Go to Google, in the search box type "site:lists.digium.com" without > the quotes then type in what you want to search for. THis will limit > all searching only to the Digium lists for asterisk. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
> forum-like tool to search thru the posts by keyworks for example.You can use google by specifying site:lists.digium.com before or after the words Most if not all of your questions are answered on the wiki (which does not seem to be responding as I write this) and at sites like http://www.asteriskdocs.org
I've just installed Astrisk with AMP. All work well but one thing is not clear. I wanna add users to allow calls between SIP phones. I've added extension but seems not to be enought. How i can add SIP users and allow calls between they ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005 www.zerozone.it
<html><div style='background-color:'><DIV class=RTE>I would like to the know following:</DIV> <DIV class=RTE> </DIV> <DIV class=RTE>1. What is the latest greatest asterisk verision? and how to get it.</DIV> <DIV class=RTE>2. can i run into with linx FC4 and kernel 2.6</DIV> <DIV class=RTE>3. how can i contribute to development of IM and Presence work on asterisk.</DIV> <DIV class=RTE> </DIV> <DIV class=RTE>Thanks</DIV> <DIV class=RTE>roswel</DIV></div></html>
_____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of roswel ajf Sent: Tuesday, January 24, 2006 10:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk newbie I would like to the know following: 1. What is the latest greatest asterisk verision? and how to get it. 1.2.2, instructions are at http://asterisk.orf 2. can i run into with linx FC4 and kernel 2.6 yes 3. how can i contribute to development of IM and Presence work on asterisk set a bounty or contribute some code Thanks roswel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060124/b86baa0a/attachment.htm