Asterisk - various questions

Vector · #1 16th August 2009, 10:48 AM

Does anyone know if there is a conference package from asterisk that is yummable. I know that there are conference programs, but i cannot find one that does not require compiling from source.

Also i noticed that when people called me via sip to sip client, that i could answer the call, but as soon as i answered it, it would disconnect. Then i noticed that if i set their clients to use a stun server at stun.ekiga.net and opened ports 10,000 to 10,024 on my firewall, that i could finally hear their voices.

I'm not sure which one is working. All of the sip clients were originally set at port 8,000 for rtp, so i'm not sure that i even needed to open the ports. Also, i was wondering if Asterisk did STUN for the clients. I do not like using a third party's stun server. So i was wondering if anyone knows how to get this to work with asterisk.

Oh, here is my setup. My computer has MANY servers, and asterisk is one of them. It is behind a firewall/router and i also use an sip CLIENT from it. The rest of the computers are different places in the world, and all behind routers. So, what would i have to do to get this working without 3rd party stun? I seen that there is a stun-server for Fedora, but i am still trying to find it's configuration files, and process name to start it up.

beaker_ · #2 16th August 2009, 11:45 AM

I'm still wrapping my mind around asterix's manual so I can't completely answer your question however I believe conferncing is done by configuring your extension(s).

An alternate is to grab dimdim, the virtual machine comes ready to roll or setup from scratch on centos 5.3 (haven't tried on fedora).

zackf · #3 17th August 2009, 05:55 PM

SIP uses port 5060, is that port open?

zackf · #4 17th August 2009, 05:56 PM

Have you tried the FreePBX overlay?

Vector · #5 17th August 2009, 05:59 PM

Yeah, the ports are open, but i don't use anything but plain asterisk and the config files.
I've got everything working, other than the nat issue and the conferencing issue.
Like i said, we can now call eachother, but third party stun is required, and there is no conferencing.

zackf · #6 17th August 2009, 06:10 PM

I thought the "MeetMe" conference was built into Asterisk. What version of * do you have?

Vector · #7 17th August 2009, 06:17 PM

the latest:

[^v^] yum list asterisk
Installed Packages
asterisk.x86_64 1.6.0.5-2.fc10

zackf · #8 17th August 2009, 06:19 PM

Does it give you any options if you type "meetme" in the asterisk cli?

Vector · #9 17th August 2009, 06:21 PM

Connected to Asterisk 1.6.0.3 currently running on Hate (pid = 2601)
Hate*CLI> meetme
No such command 'meetme' (type 'help meetme' for other possible commands)
Hate*CLI> help meetme
No such command 'meetme'.
Hate*CLI>

zackf · #10 17th August 2009, 06:34 PM

Interesting, are you using zap or dahdi for your outside lines?

Vector · #11 17th August 2009, 06:55 PM

I have no outside lines. I have no hardware for pots. I have a phone number provided to me by google, and i forward it to my sip server when someone calls it; it works fine.

All users who use sip just use twinkle, because ekiga sux.

Every article i read for conferencing says that you have to have zapta hardware for the timing clock (or some kind of zapta dummy driver), but i don't use pots at all; everything is sip from end to end.

zackf · #12 18th August 2009, 06:41 PM

That's pretty clever, You do need a timing source, but you can compile "dummy" interfaces to do this. However, I don't think any of them are yummable. If you are interested in Asterisk, FreePBX has come a really long way and has a super easy interface.

http://www.freepbx.org/support/docum...for-centos-5-1

Is a great guide, should be very similar for Fedora.

When you compile dahdi you should have access to the dahdi_dummy timing source.

Vector · #13 18th August 2009, 07:03 PM

I try to stay away from compiling anything, as any time you update your kernel, you have to recompile it, meaning that you also have to keep a copy of the source, and that's assuming no dependencies were broken in the update.

If i can't yum it, i do without.

That may sound a bit pious, or whatever, but it can sometimes be an advantage. For example, when Cinelera stopped packaging and trying to be compatible with fedora, without fourth party repos (back then, if it was not Livna or Fedora, i would not use it), i stopped trying to install and learn it. As a result, i had to learn command line video editing, and i'll swear that is the best way to edit any video. As a result, I have transcoded over 300 DVDs into high quality mkv/h264/aac files.

zackf · #14 18th August 2009, 07:14 PM

Asterisk does ok through updates actually, but the drivers on my PRI cards to need recompiled if I yum update. Since my PBX's are all internal I generally don't need to update them once they are off and running.

Vector · #15 18th August 2009, 08:52 PM

Do you use broadband.com for your host, or do you host from your local machine? I never bought a DID. I wound up getting an invite for Google Voice, and they gave me a free number (100110010000110010111010111000001), which i have to forward to Gizmo (free, for forwarding only), and then to my local SIP server. I'm still considering a DID. I actually found a company that offers Global Unlimited for 30$/mo.

Also, do users ever connect to your machine via SIP CLIENTS, or are your peers using PBX systems also? I'm still a bit confused about the need for a stun server, because there is an rport header, and i believe another header that an SIP client will send to the PBX server, which is normally used, and i think that if this is configured properly, that there is no NAT issue. In other words, i think that i have a misconfiguration on my end.

I'm guessing that you already know how to do your hold music. I'd offer the files i have, but i doubt that you're into metal (Devildriver, Violent Work of Art, Fear Factory, etc)

. Besides, i still need to reduce their amplitude, as they are too loud for telephones and sound very bad (SIP clients are fine though).

Thanx