Asterisk Setup for noob...

[dswartz]

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

[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!

[NY Tel Guy]

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!

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

[scott2020]

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

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

[zevin]

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.

[zevin]

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.

[danielbeck]

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)

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

Here is a log of one of the recent attempts:

[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'

[danielbeck]

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):

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