Re: [asterisk-users] Beginning to use Asterisk and tests with extensions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El domingo 10 de mayo del 2009 a las 17:12:51 -0300, Daniel Bareiro escribió: >> I suggest testing your SIP softphone with the Echo() and/or >> Playback() dialplan applications before attempting to call another >> softphone/hardphone/etc. This will allow you to confirm that the >> one endpoint functions properly before adding more complexity by >> calling another endpoint. > I was testing and sometimes with Echo() and MusicOnHold the sound is > broken. Is there some form to solve this? Investigating a little more in Internet, it seems that the expression used in english for this is "choppy sound". According to it seems, the problem was of the side of the client and I could solve it of the form commented here [1] (softphone I use is Twinkle). The problem with this is that although Twinkle now work, I can't have anything else running that uses sound because then the audio is blocked. Regards, Daniel [1] http://www.lynchconsulting.com.au/blog/index.cfm/2008/11/6/Choppy-sound-on-Twinkle-Softphone-on-Ubuntu-Linux -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoHX1UACgkQZpa/GxTmHTezugCdFqm+vBOIwsTGUYS7ecTT7DcP TX4An0kU/YtK5w1CDMEln8vabOzRJNC4 =iaWE -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginning to use Asterisk and tests with extensions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 > Hello Daniel, Hi Dana. > You will find the information at http://www.voip-info.org/ and > http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the > "Online Book" link) very useful. I have the second edition that covers Asterisk 1.4 and it seems interesting. You made me remember that I had downloaded it the last year, although just now I have more time to dedicate to Asterisk. The fact of to have already installed it is an important step :-) > The asterisk package by itself should be adequate for SIP/IAX calls. > I don't think you need libpri unless you are planning on connecting asterisk > to a digital connection such as ISDN or a PRI. > You will need Zaptel (for Asterisk versions 1.2,1.4) or DAHDI (Asterisk > versions >=1.6) if you choose to install an internal card (OpenVOX, Digium, > Sangoma, etc.) I do not know if or how well this will work with a VM. Thanks for the indication. According to I saw in the site of Asterisk[1], only make reference to DAHDI for Asterisk 1.4, but according to which you say to me, both can be used. My idea is to buy an ATA to connect a conventional telephone and make tests of communication between it and softphone. The idea by which I thought about using an ATA is because I am not sure with my version of KVM (KVM-62) can make PCI pass through. But with the ATA must not have problem. Having this in mind, I installed the packages dahdi-linux-2.1.0.4.tar.gz and dahdi-tools-2.1.0.2.tar.gz having loaded only the module dahdi_dummy and so far commenting all that appear in /etc/dahdi/modules. > I suggest testing your SIP softphone with the Echo() and/or Playback() > dialplan applications before attempting to call another > softphone/hardphone/etc. This will allow you to confirm that the one > endpoint functions properly before adding more complexity by calling another > endpoint. I was testing and sometimes with Echo() and MusicOnHold the sound is broken. Is there some form to solve this? > some things that allow you to call a conventional telephone: > an ATA with an FXS port > an internal card (such as OpenVOX, Digium, Sangoma) with an FXS port > call a conventional phone number through the PSTN (below) > > To connect to the PSTN you can use any of: > an ATA with an FXO port (plug an analog phone line into it) > internal card with an FXO port (also to plug an analog phone line in) > account with an ITSP (there is occasionally discussion on the list about > advantages/issues/opinions/and flames with various ITSPs - google > "site:lists.digium.com ITSP") > [...] I believe that with the example I understood a little better how it works. As it mentioned above, I am thinking about buying a Linksys SPA3102 to make both internals and with PSTN tests. > Hope that gets you going in the right direction. > > http://www.voipsupply.com/ is a good source to see what equipment is > generally available to end users. Thanks for your reply and by all the references and examples that you provided to me. Regards, Daniel [1] http://www.asterisk.org/downloads -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoHNXIACgkQZpa/GxTmHTccUACfbHY+st10rhsqrsZnE9SJLZrV hFQAnR5Y85XQQr7Jm1wWzD106qxNkd4g =EYud -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginning to use Asterisk and tests with extensions
Hi there Daniel,
I havnt caught the beginning of this thread - but as we speak I'm in the
process of installing an Ubuntu server into a VM and getting 1.4 up and
running in a hope to replace my home system.
I did kinda follow a link
http://support.red-fone.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=13
that goes through a few extra libraries that I tried to set up from the
outset that didnt - (I'm still learning linux as well). I had to adjust
the settings for '/apt-get install linux-headers-2.6.24-19-server'/ to
reflect the version of Ubuntu server ('cat /proc/version')
At this point I've been crawling through the O'Reilly link (I've
actually got the 1st edition in real print(!) but this ones geared to
v1.4. Asterisk at this point is running - even to the degree I can get
the web page GUI up (I've installed that too).
So far though - no phones or lines - still getting there - but Asterisk
'seems' to be running - albeit VERY idle at this point!
Cheers
Wayne.
Dana Harding wrote:
> Hello Daniel,
>
> You will find the information at http://www.voip-info.org/ and
> http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the
> "Online Book" link) very useful.
>
>
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Re: [asterisk-users] Beginning to use Asterisk and tests with extensions
Hello Daniel, You will find the information at http://www.voip-info.org/ and http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the "Online Book" link) very useful. The asterisk package by itself should be adequate for SIP/IAX calls. I don't think you need libpri unless you are planning on connecting asterisk to a digital connection such as ISDN or a PRI. You will need Zaptel (for Asterisk versions 1.2,1.4) or DAHDI (Asterisk versions >=1.6) if you choose to install an internal card (OpenVOX, Digium, Sangoma, etc.) I do not know if or how well this will work with a VM. I suggest testing your SIP softphone with the Echo() and/or Playback() dialplan applications before attempting to call another softphone/hardphone/etc. This will allow you to confirm that the one endpoint functions properly before adding more complexity by calling another endpoint. some things that allow you to call a conventional telephone: an ATA with an FXS port an internal card (such as OpenVOX, Digium, Sangoma) with an FXS port call a conventional phone number through the PSTN (below) To connect to the PSTN you can use any of: an ATA with an FXO port (plug an analog phone line into it) internal card with an FXO port (also to plug an analog phone line in) account with an ITSP (there is occasionally discussion on the list about advantages/issues/opinions/and flames with various ITSPs - google "site:lists.digium.com ITSP") An ATA (Analog Telephone Adapter) is basically an analog to digital converter. CISCO/Linksys does manufacture some ATAs, but this is not the only option for an ATA. There are two types of analog ports - FXO, and FXS. You plug a telephone into an FXS port. You plug a conventional phone line into an FXO port. Bad things can happen if you plug a phone line into an FXS port - do not do it. In your example of an ATA with two FXS ports, you will use two conventional telephones - one plugged into each port. The ATA will use a different user account for each of the analog ports. This requires configuring each user account on both the ATA and in Asterisk's sip.conf. The ATA functions simply as a passthrough device: "phone calls for user1 = ring port1" "phone calls for user2 = ring port2" "phone calls from port1 = use user1 account" "phone calls from port2 = user user2 account" The ATA does not decide which port is connected to which extension number. This actually happens in Asterisk's dialplan. A very basic extensions.conf to illustrate different extensions to call each port, and one to call both: [default] exten => 101,1,Dial(SIP/user1) exten => 102,1,Dial(SIP/user2) exten => 103,1,Dial(SIP/user1&SIP/user2) Hope that gets you going in the right direction. http://www.voipsupply.com/ is a good source to see what equipment is generally available to end users. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Beginning to use Asterisk and tests with extensions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! This is my first message to the list/newsgroup. This weekend and after to have fought by some time with my soundcard with respecto to the voice capture, after assuring to have solved that problem, I installed Asterisk on Debian GNU/Linux Lenny. I made my installation on a KVM virtual machine. In order to begin and according to I could see on the basis of which I was reading in Internet, to make a basic installation initially it would be enough with the packages 'asterisk' and 'libpri', reason why those were these that I installed at the moment. But correct to me, if I'm mistaken, please. However, the following basic step would be to test with extensions and since in my house I only have a PC that use like workstation, is some "complicated" to test of calls :-) Whatever happens, I installed Twinkle from Debian GNU/Linux repositories. But to make a valid test would need another PC with softphone or "something" that allows me to call to a conventional telephone. For this I, read in some documents that the ATAs are mentioned (bah, I believe that the denomination "ATA" is something own of CISCO and perhaps most appropriate is to call it as Adapters for Analogical Telephones), that allows to connect a conventional telephone to a VoIP network of way to be able to send and to receive calls having an Ethernet connector to connect it to the LAN. What not yet it is clear to me of these ATAs is how they works. I have understood that it have its own IP and the one of PBX server, but if we have, for example, two FXS ports connecting to each of them to a conventional telephone, in the documentation that I found at the moment is not mentioned some way to associate the ports of the conventional telephones with a number of extension so that the ATA knows how to route an incoming call. The other alternative is to use a OpenVOX card, for example, but I'm not sure if this solution is worth to me because if I install it in the PC where I have the virtual machine with the Asterisk, I'm not sure if the KVM virtual machine can access to that underlying hardware. Thanks in advance and with the time I hope to be gaining knowledge also to be able to make some contribution to the list/newsgroup. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoAD3QACgkQZpa/GxTmHTeL/QCZAa0cy77+5YIHAx5wRvDeSvTd 01EAn2yPyHuyeaabkqLL9yW4S0rb2USx =2TBa -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
