I completely don't get it

I have followed all the examples in the Oreilly book and Packt book and various examples online, and I simply can’t get this sh1t to work. Everytime I get to the point in every example where is says do this and this and this and it’ll work, it simply doesn’t. I don’t even know where to start. I have no idea why it’s not working, I have no idea where to look. I’ve looked in /var/logs, I’ve looked on the console, I’ve looked in the logs for the devices I’m am trying to connect and get nothing, zero, nada. All I am trying to do, is setup Asterisk to use SIP. I have no idea why this isn’t working. It all seems pretty straight forward, but when I follow the directions, I don’t get shit.

Sorry for the rant, but I am just super pissed off and I have spent hours with this thing and have nothing to show for it. If anyone can tell me where to start, send me your contact info and hourly wage, and we’ll go from there…

You should start there : yogalearningcenter.com/
Then read the Oreilly Book completely before playing with Asterisk.

hang on hang on. first, asterisk has a steep learning curve. You are not the first one to be banging their head into a wall.

Second, what exactly do you want * to do (in general)? What kind of lines do you have and what do you want to do with the calls that come in on them? What phones do you have? How many?

Answer these and I will help you if I can…

Tell us more specifically what it is you’re seeing and we’ll try to help.

There are some really simple dial plans in the O’Reilly book. Try them out first and build on them only when you’ve got the simplest ones working.

My first few dozen attempts at getting Asterisk to do anything useful were frustrating, to say the least. Stick with it, it starts to make sense after a while.

A great way to start learning asterisk IMO is by using trixbox…
www.trixbox.org

Then once you are familliar with it, you can move to a custom built system.

i’d actually go back a stage and say A@H is better

/dons flame suit.

Thanks for the responses everyone. Again, sorry for the rant, just needed to let off some steam. Thanks for the advice Dimitripietro, I’m sure I could use it.

Now, the problem: I am trying to set up a small office (i.e. 25 extentions). I brought home a couple of the phones and a server to play with. I am running Asterisk 1.2.7.1 on Fedora 5 2.6.17-1.2159_FC5. I am trying to register 2 GXP-2000 phones and X-lite Version 3 bld 30942. This is my ‘lab’ setup if you will. After thinking about it, I have not been able to get a phone to register this entire time. Neither the GXPs or the Xlite. For all I know, I have Asterisk setup correctly, but don’t know since I can’t get a device registered to test it. All I know is that when I type ‘sip show peers’, there is nothing except the junction networks account I setup, which seems to connect, but again I have no way of testing it.

Here are the configs for the phones:

Xlite:

GXP:

Here is my current sip.conf (mind you this has gone through many revisions):

[general]
context=default
srvlookup=yes
port=5060
bind=0.0.0.0

[john]
type=friend
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=sip

[phone1]
type=friend
username=blah
secret=blah
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=sip

[phone2]
type=friend
username=blah
secret=blah
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=sip

And extensions.conf:

[sip]
exten => 1,1,Dial(SIP/phone1,20,tr)
exten => 2,1,Dial(SIP/phone2,20,tr)
exten => 3,1,Dial(SIP/john,20,tr)
exten => 1000,1,Dial(SIP/phone1&SIP/phone2&SIP/john,20,tr)

I don’t have a zapata.conf or an iax.conf, nor do I need them right???

The conf’s are real simple because I was trying to eliminate an potention interference from example confs and keep in mind that the sip.conf and extensions.conf have gonna through numerous changes and revisions. They may not look like what was in the book because I have tried different examples from many different sources.

Please let me know if there is anything else I can tell you all, I just want to get this thing working and again, thanks for any and all help!

If you need anything, I will be in the corner performing the Downward Facing Dog Pose, thanks. :wink:

Also, there was some mention of using trixbox. Would that fly in a ‘professional’ setting?

Trixbox would be fine in the setting you describe. It packages * , freePBX etc into an iso complete with CentOS. It will format your harddrive and take over, but throw a spare hd in and save yourself some grief. Follow the Nerdvittles guide below and you will have something that actually works to learn on. Then you can either use that as a reference to set up * from scratch at work or you can use the Trixbox setup as your base.

Here’s a nice guide to setting up Trixbox nerdvittles.com/index.php?p=137

Here’s something tailored to a small business Trixbox setup sureteq.com/asterisk/trixbox.htm

Ok, I downloaded the iso, thanks for the info George. I’ll play with that a little later. In the meantime, I would like to try and salvage my * setup if anyone feels it’s salvagable. Thoughts?

one immediate problem i see- xlite should be registering by proxy, NOT domain.

on the GXP leave outbound proxy blank, as well as auth id (usually)

where you have defined [phone1] must be the same as SIP user name! This is probably why you are having trouble. Set the SIP user name on the GXP to be phone1 and change username in the sip thing to be phone1 too (keep it simple).

As for A@H, its something to think about but it may not be for you. It’s like an M&M except the candy shell is two inches thick and you need a pickaxe to get to the chocolate center. It’s a great way to get * up and running quickly, but IMHO it’s a bad way to learn about *. The dialplans it creates are very large, and there is a full page of debug data for a single call (because it goes through all its features and conditionals for each call).

Ok, here is how my sip.conf looks now:

[general]
context=default
srvlookup=yes
port=5060
bind=0.0.0.0

[john]
type=friend
username=john
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=sip

[phone1]
type=friend
username=phone1
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=sip

[phone2]
type=friend
username=phone2
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=sip

I configured xlite with the following:

It errors with 'Registration Error: 408 - Request Timeout

Here is the log:

http://subnetsystems.com/asterisk.log.txt

I configured the GXP-2000s like this (except phone1 has ‘phone1’ where applicable:

but neither of them register.

After doing a sip reload and bouncing all devices, a sip show peers looks like this:

tempwork*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
phone2/phone2 (Unspecified) D 0 UNKNOWN
phone1/phone1 (Unspecified) D 0 UNKNOWN
john/john (Unspecified) D 0 UNKNOWN
3 sip peers [0 online , 3 offline]

Any ideas what could be wrong?

OK start simple…grab a IAX softphone here laser.com/dante/

create IAX exten

Is there a firewall on the asterisk server??? open port 4569 if so.
on a PC on the LAN run the softphone diax.exe and then setup the exten single page setup easy

Alias… Use the extn number
Username… is EXTN
Password …password
leave context blank
put tick in Register
save it
if it lights the number 1 green then u are connected…dial *72 hear anything???

Sip hates firewalls…hates DUMB smart routers (SPI)

ports are UDP not TCP for SIP…

Cisco says "Server could not produce a response before the expiration timeout."
looks like some routing problem ?
do You have switch or router with firewall to connect sip devices/computer with * box ? firewall on * box ?
Can You ping * box from computer where x-lite is installed ?
what is ping result ?
i have in home very similar configuration, my sip.conf looks almost the same as Yours
in my x-lite conf i left blank Auth. user name filed, domain field is my * box ip adress

Um…heh…yeah, uh ya see in addition to being an * noob, I’m still a bit of a linux noob too, and…uh…heh, yeah well I didn’t realize that a software firewall got installed, so um…I didn’t really check that. Everything is fine, thanks for all the help, shows over nothing to see here. I’m just going to disappear into that dark corner over there where no one can see me… :blush: :blush: :blush:

But thanks again for all who helped, sorry my introduction to this board was a little hot headed…

The fact that you can admit means that you’ll make it just fine. It takes a sense of humor to be involved in this stuff. Well, that and a WHOLE lot of patience.

I spent all day upgrading the firmware on a Cisco phone from an old SIP firmware to SCCP. Did I mention it takes a WHOLE lot of patience? It was nobody’s fault but my own. It’s usually the little things that get you.

I haven’t messed with SIP in a few weeks, and I was also trying to get a SIP phone working at the same time. This thread was a refresher. I forgot that the SIP definition has to be the same as the username.