Asterisk Setup for noob...

Ok, here is one for you all to ponder for today. I am green and would like to play and learn.http://img108.exs.cx/img108/2817/m8ztelephone.gif

I have an older (2 years old) compaq installed with asterisk@home or asterisk now 1.0.2 (confused over all the names). Installed it all last night with no problems.

* is connected to my network via router then modem. I have a linksys wireless 4 port router. I set up a static ip in router via DHCP and Mac address. Machine now connects to 192.168.1.111 no problem.

I was not able to connect to the * via 192.168.1.111 in web browser. I rebooted * and performed line by line verification of boot. I did not authorize the firewall this time and was able to connect to * via web browser.

Now I rebooted the * last night and have not been able to connect to * GUI.

How would I modify the firewall to allow access. I will also need some help further along. Will I be able to play with this using the service VOIPo provides? Do you guys have an * running and how do you use it?

I would ultimately like to use this as a PBX on my property to call the apartments in the back. I have a few apartments for college students on my property. They all have cat5 drops.

Thanks for your advise and tips. I am really green and am eager to learn.

If this machine is behind the router and needs no firewall, I think (I use redhat, which is what I think trixbox uses) you want this:

chkconfig firewall off

p.s. do yourself a huge favor here: what you are running is OLD. chock-full of bugs and honestly it sucks. a good distro to try is pbxinaflash.

Quote:

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Originally Posted by dswartz

If this machine is behind the router and needs no firewall, I think (I use redhat, which is what I think trixbox uses) you want this:

chkconfig firewall off

p.s. do yourself a huge favor here: what you are running is OLD. chock-full of bugs and honestly it sucks. a good distro to try is pbxinaflash.

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I did not know that... There seems to be so many variations to the * system.

I am open to any and all suggestions. I will look at pbxinaflash and I will probably install tonight. Anything I should be aware of?

Thanks for your reply dswartz, keep the ideas coming.;)

Not offhand. I've not used pbxinaflash, but I've heard good things and it is less flaky than trixbox.

PBX In a flash is great and I think much cleaner than trixbox. They are also building it so upgrades aren't a huge pain. There are lots of cool scripts made for it too that install neat little extras.

However, in another thread about odd disconnects, there is some discussion about Asterisk 1.4 with VoipO disconnecting after 30 minutes. It apparently has something to do with session timers not being supported in Asterisk 1.4.

But, the PIAF ISO image allows you to install 1.4 or 1.6 beta, and the 1.6 beta does work with session timers. I haven't tried it yet personally but I plan to.

Scott

I can confirm the session timer aspect. I am running asterisk on a separate box (not pbxinaflash) and migrated from 1.4 to 1.6 so I could make voipo calls longer than 30 minutes :)

I love the help and chatter, keep it going.

I tried to install the pbxinaflash last night but was unable to get the CD to run at bootup. I checked and the CD drive is number one in boot order. I was using a CD-RW would that make a difference?

I am going to try again tonight with a new copy. Any ideas as to why the ISO that I burned would not boot to install.

no idea, maybe downloaded or burned it wrong?

Quote:

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Originally Posted by zevin

I love the help and chatter, keep it going.

I tried to install the pbxinaflash last night but was unable to get the CD to run at bootup. I checked and the CD drive is number one in boot order. I was using a CD-RW would that make a difference?

I am going to try again tonight with a new copy. Any ideas as to why the ISO that I burned would not boot to install.

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Since I have forgotten to do so at times, be sure the ISO you burn is set up to be burned as a bootable ISO.

FWIW I use a free program on Vista http://www.imgburn.com/ that is great for ISO distros.

I don't expect you would have just copied the data to a CD or DVD anyway - just thought it is a good plug for the software :)

I was able to burn the ISO just fine using Nero. The only blank CD I had was a CD-RW. I was excited and couldn't wait any longer so I use the CD-RW. I think that is my problem. For what ever reason it doesn't like the CD-RW.

I am now burning again to a CD-R. I also finally figured out how to get GRUB to stop taking over and allow the CD to boot. BIOS was set to CD Rom drive as the first but GRUB would overpower BIOS.

I hope tonight I will see some progress.

For what ever reason the CD-RW would never boot.

I like DeepBurner for burning ISO images. They have a free version. In the past, I have had trouble sometimes with an ISO burn that wouldn't boot, only to re-burn it to another CD and it works fine. My problems have always been dirty CD drives and stuff like that.

I installed PbxIAF on a virtual machine last night with the Asterisk 1.6 option. I could not for the life of me get into FreePBX though. It kept asking for the admin and maint passwords, and I kept entering them, only to get rejected. I went ahead and updated to FreePBX version 2.5 hoping to fix it, but no real luck there. I ended up going into the amportal.conf (I think) file and taking out the database login check thing, and removed the .htaccess from the /admin pages and I did get in. But, I do get hammered for passwords still and I just keep entering in stuff until it lets me in! Good grief.

Tim is sending me a Grandstream that I will use for now. I'll play with 1.6 and move everything over to it eventually.

Scott

It worked! I got it installed. Quite a few steps. It took a little longer than I thought, before I knew it the time was 3:30 in the morning and I had to work at 9.

So tonight I get to play!

I will keep you guys posted. Thanks for the help

Oh and the CD-R worked much better.

Quote:

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Originally Posted by zevin

It worked! I got it installed. Quite a few steps. It took a little longer than I thought, before I knew it the time was 3:30 in the morning and I had to work at 9.

So tonight I get to play!

I will keep you guys posted. Thanks for the help

Oh and the CD-R worked much better.

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I had an employee who worked for me once and his name was Devin, his brother was bevin and there was a third son with a rhyming name. True story.
Perhaps you are the missing link.;)

Did you end up going with 1.4 or 1.6?
So far Asterisk 1.6 with FreePBX 2.5 seems pretty slick. I need to play some more with it, but after the wife goes to sleep or something.. :)

Quote:

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Originally Posted by scott2020

Did you end up going with 1.4 or 1.6?
So far Asterisk 1.6 with FreePBX 2.5 seems pretty slick. I need to play some more with it, but after the wife goes to sleep or something.. :)

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I played it safe and installed the 1.4. I figured I had enough to learn with out dealing with beta problems. I will probably upgrade, I am gluten for punishment. I feel the best way to learn is to get down and dirty and not to be afraid to break it. You can always reinstall.

I got "her" going, updated and all. I have not installed a Trunk yet. What settings have you all found to work here? tweakings? Will I be able to use sip.voipwelcome.com? I assume so, like I said I am learning this all.

I have installed a softphone and was able to play with the voice mail and got that all jazzed up- with weather what a dork I am. I just have not attempted to place outside calls. I will tonight!!!

Quote:

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Originally Posted by NY Tel Guy

I had an employee who worked for me once and his name was Devin, his brother was bevin and there was a third son with a rhyming name. True story.
Perhaps you are the missing link.;)

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It could be true, I am adopted... Did they just work through the alphabet?

"Honey I just loved how Bevin sounded lets name our next Cevin oh and Devin and when we get to Zevin everyone will know we are nuts."

The physical education teachers had a good time with me in high school.

Quote:

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Originally Posted by zevin

I played it safe and installed the 1.4. I figured I had enough to learn with out dealing with beta problems. I will probably upgrade, I am gluten for punishment. I feel the best way to learn is to get down and dirty and not to be afraid to break it. You can always reinstall.

I got "her" going, updated and all. I have not installed a Trunk yet. What settings have you all found to work here? tweakings? Will I be able to use sip.voipwelcome.com? I assume so, like I said I am learning this all.

I have installed a softphone and was able to play with the voice mail and got that all jazzed up- with weather what a dork I am. I just have not attempted to place outside calls. I will tonight!!!

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Yes, sip.voipwelcome.com works fine. My inbound and outbound for voipo:

Outbound:

username=NNNNNNNNNN
fromuser=NNNNNNNNNN
fromdomain=codeblue.voipo.com
type=peer
secret=XXXXXX
qualify=yes
host=sip.voipwelcome.com
disallow=all
allow=ulaw
insecure=invite

Inbound:

type=peer
secret=XXXXXX
qualify=yes
insecure=very
host=sip.voipwelcome.com
disallow=all
allow=ulaw

Can I resist? :p
From 2004: http://www.dslreports.com/forum/remark,9193232

Quote:

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Apparently, 1 in 5 people in the world are Chinese. And there are 5 people in my family, so it must be one of them.

It's either my mom or my dad... or, maybe my older brother Colin or my younger brother Ho-Cha Chu. But I'm pretty sure it's Colin.

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Well I can call out but am unable to call in to the box. I see were the device is registered in the vpanel. I have a dinamic IP and am using DynDNS. I am at a loss for ideas.

I will post my inbound config when I get home tonight.

why post your outbound config is it's inbound that's broken?

Quote:

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Originally Posted by dswartz

why post your outbound config is it's inbound that's broken?

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That's what I get when I don't proof my message before sending.:eek: I have fixed my error thanks for pointing it out to me.

I am also on dynamic IP with dyndns and it seems to be OK for me. I abandoned my virtual machine because it was too quirky. It was messing up a lot.

I had to edit my sip_additional.conf and put in my externhost=mydyndnsthing.dyndns.org and the other NAT related things there. Also opened up the ports on the firewall and such. Not sure if you already did this but I can post my stuff if you haven't tried that.

scott

No, do NOT edit sip_additional.conf! You want sip_custom.conf - any non-custom files may be overwritten without warning!

Thanks guys, I was not able to play with my machine last night as my 2nd half was on a rampage the second I pulled into the driveway. Something about the dog, how I forgot to stop at the store for milk and TP, my clothes being on the bathroom floor, and something about not paying attention. It was just crazy at home last night so I didn't dare drag out the machine.

I really need a man shed to run to when things get hot. One with A/C, Internet connection, cable, a small kitchen, bathroom. Hell I should just rent an apartment.

Better luck tonight!;)

Quote:

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Originally Posted by zevin

Thanks guys, I was not able to play with my machine last night as my 2nd half was on a rampage the second I pulled into the driveway. Something about the dog, how I forgot to stop at the store for milk and TP, my clothes being on the bathroom floor, and something about not paying attention. It was just crazy at home last night so I didn't dare drag out the machine.

I really need a man shed to run to when things get hot. One with A/C, Internet connection, cable, a small kitchen, bathroom. Hell I should just rent an apartment.

Better luck tonight!;)

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Sounds like you need to move out......:eek:

Quote:

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Originally Posted by dswartz

No, do NOT edit sip_additional.conf! You want sip_custom.conf - any non-custom files may be overwritten without warning!

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My bad!
Sorry about that. It is the sip_custom.conf. If you are using PIAF it only shows you files that are safe to edit, AFAIK.

My better half pretty much got fed up when the 30 minute drops started happening. Then I started with the Asterisk 1.6 thing, and finally had to get the VoipO Grandstream. The Grandstream just works, and works pretty well. It locked up on me once, but since has been great. That will give me some breathing room to play with 1.6 a little...

Scott

Ok, guys... Got the inbound and outbound working... I had played with the incoming route rules and that was preventing the calls from coming in. Silly me.;)

How can I allow incoming callers the ability to get the weather or other options I have installed. I have installed the zip code weather mod. I have it setup so extensions can dial 947 to retrieve weather. But I would like to give incoming callers the ability to check the weather also.

Perhaps I should post this on the PiaF forum, I just thought you guys would know.

Found it... I am looking for an auto attendant.

Introducing the Stealth AutoAttendant for Asterisk 1.4 and FreePBX

I am becoming a fan of Nerd Vittles.

Zevin,

I'm now where you were a while ago and I can't seem to get inbound calls setup properly. I have setup an inbound route that will ring all phones (call group 600) and I've tried numerous settings in the Incoming Settings in the SIP Trunk settings section. Outbound calls work fine.

Will someone with a working freePBX setup please post their inbound settings so I can compare what I have with what others have?

I know the server is getting the calls but depending on how I have the inbound settings configured, I either get what sounds like dead air (no ring, no voice, nothing) or I get the recorded
attendant telling me the number I have dialed is not in service.

Here are my current User Details: (note: I have User Context set to my VOIPo number)

Code:

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type=peer
secret=XXXXXXXXX
qualify=yes
insecure=very
host=sip.voipwelcome.com
disallow=all
allow=ulaw

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Here is a log of one of the recent attempts:

Code:

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[Oct 26 11:50:45] VERBOSE[2895] logger.c: == Using SIP RTP TOS bits 184
[Oct 26 11:50:45] VERBOSE[2895] logger.c: == Using SIP RTP CoS mark 5
[Oct 26 11:50:45] VERBOSE[2895] logger.c: == Using SIP VRTP TOS bits 136
[Oct 26 11:50:45] VERBOSE[2895] logger.c: == Using SIP VRTP CoS mark 6
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Executing [3853515383@from-sip-external:1] NoOp("SIP/VOIPo-0000005f", "Received incoming SIP connection from unknown peer to 3853515383") in new stack
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Executing [3853515383@from-sip-external:2] Set("SIP/VOIPo-0000005f", "DID=3853515383") in new stack
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Executing [3853515383@from-sip-external:3] Goto("SIP/VOIPo-0000005f", "s,1") in new stack
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Goto (from-sip-external,s,1)
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:1] GotoIf("SIP/VOIPo-0000005f", "0?from-trunk,3853515383,1") in new stack
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:2] Set("SIP/VOIPo-0000005f", "TIMEOUT(absolute)=15") in new stack
[Oct 26 11:50:45] VERBOSE[15415] logger.c: Channel will hangup at 2011-10-26 11:51:00.000 PDT.
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:3] Answer("SIP/VOIPo-0000005f", "") in new stack
[Oct 26 11:50:45] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:4] Wait("SIP/VOIPo-0000005f", "2") in new stack
[Oct 26 11:50:47] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:5] Playback("SIP/VOIPo-0000005f", "ss-noservice") in new stack
[Oct 26 11:50:47] VERBOSE[15415] logger.c: -- <SIP/VOIPo-0000005f> Playing 'ss-noservice.gsm' (language 'en')
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:6] PlayTones("SIP/VOIPo-0000005f", "congestion") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:7] Congestion("SIP/VOIPo-0000005f", "5") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/VOIPo-0000005f'
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [h@from-sip-external:1] NoOp("SIP/VOIPo-0000005f", "Hangup") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [h@from-sip-external:2] Set("SIP/VOIPo-0000005f", "DID=s") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [h@from-sip-external:3] Goto("SIP/VOIPo-0000005f", "s,1") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Goto (from-sip-external,s,1)
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:1] GotoIf("SIP/VOIPo-0000005f", "0?from-trunk,s,1") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:2] Set("SIP/VOIPo-0000005f", "TIMEOUT(absolute)=15") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: Channel will hangup at 2011-10-26 11:51:08.000 PDT.
[Oct 26 11:50:53] VERBOSE[15415] logger.c: -- Executing [s@from-sip-external:3] Answer("SIP/VOIPo-0000005f", "") in new stack
[Oct 26 11:50:53] VERBOSE[15415] logger.c: == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/VOIPo-0000005f'

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So, after much research, I was able to find the proper trunk config setting that would work with VOIPo. You can find them in this thread: http://forums.voipo.com/showthread.p...isk-Help/page2

It turns out that the way VOIPo is setup. The incoming settings (USER context and USER details) aren't even used since they expect users to be using a VOIP adapter therefore you have to establish the context in the PEER details section. In case the above thread disappears for some reason. Here are the settings that worked for me (put these in the PEER Details):

Code:

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username=38XXXXXXX
type=peer
session-timers=accept
session-refresher=uac
secret=XXXX(Put your SIP password here)XXXXX
rfc2833compensate=yes
qualify=5000
nat=no
insecure=port,invite
host=sip.voipwelcome.com
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw

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