Asterisk@home (for Business)

The Clown Man

Right, inbound calls are hitting my asterisk now but my CLI is reporting

-- Executing AbsoluteTimeout("SIP/Ahernmcdonnell-29d6", "15") in new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/Ahernmcdonnell-29d6", "") in new stack
== Spawn extension (from-sip-external, 14, 2) exited non-zero on 'SIP/Ahernmcdonnell-29d6'
-- Executing AbsoluteTimeout("SIP/Ahernmcdonnell-29d6", "15") in new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/Ahernmcdonnell-29d6", "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/Ahernmcdonnell-29d6'

I have incoming calls forwarding to ext 14 (me) but I'm not recieving any calls. I also have /14 at the end of my register string ... but tbh I'm still not entirely sure if I'm getting the register string right.

Anyone have any idea what may be wrong?

The Clown Man

Also,

I've narrowed the voicemail problems down to being a codec problem.

For some reason I cannot hear sound from my voicemail. The call connects to *77 no problem. "show g729" tells me that 1 decoder is being used but no encoders are being used when the call is active.

So sound to the voicemail is using g729 but sound from is not.


Can anyone help me with this cause I'm at a loss as to what to do. Google is getting tired of me.

The Clown Man

Voicemail fixed now.

Codec installed was for the wrong processor. There is a g729 codec for each processor which I would have seen if I had have read the README properly!

Also,

Jaden,

I've come across this which seems to be a better QoS fix than traffic shaping for IPCop routers. I haven't got it fully working and I have not tested it yet but between this and QoS management on the above mentioned switches I may be able to set up a decent QoS solution for VoIP over a shared LAN. I'll keep you informed on how I get on.

antoinolachtnai

Hang on, let me understand this - how are you going to segregate the phones from the PCs if you are going to connect the PCs to the ethernet _through_ the phones? Or do the phones have vlan support which you plan to use?

The Clown Man

They seem to support vlan.

Main Features Multiline support, integrated Ethernet switch, Power over Ethernet (PoE) support
VoIP Protocols SIP
Voice Codecs G.711, G.722, G.723, G.728, G.729, EFR
Quality of Service IEEE 802.1Q (VLAN), Differentiated Services (DiffServ), IEEE 802.1p, Type of Service (ToS)
IP Address Assignment DHCP, PPPoE

That and the QoS on the router should be well enough to keep things stable. I have ordered the switch and a grandstream phone to do some setup and testing.

The Clown Man

Btw you are on 666 posts.

The Clown Man

Just to update:

I seem to have the QoS issues sorted as far as I can see. Even on my current 4MB/384KB dsl line I can hold a completely unaffected and perfectly clear g729 call while I am uploading and downloading to max capacity and xferring 4 iso files internally including one from the PC that is running through the grandstream loop and into the switch. I'll describe the setup below:


Grandstream GPX-2000 using 802.1p QoS on priorty 7.

SMC Smart Switch with 802.1p tagging enabled.

Ipcop running 802.1p QoS addon with "layer7 rule" set to catch SIP data and tag priority 7 for routing.


So by just using 802.1p QoS on my router, switch and phones I have every SIP packet up to my isp's backhaul prioritised as far as I can see.

Also, I've had to ditch the Metro idea and I will be going with IBB for a 4MB/4MB 8:1 line at roughly the same price.

From the well documented issues that people have been having with IBB I was originally reluctant to take a line with them. So, I made it clear that the latency on the available bandwidth needs to stay below 50ms regardless of contention. (It has now been written as a condition of the contract.) Also, I asked them to prioritise RTP traffic which SIP uses. They readily agreed to it and it has been put into effect now.

I have to say that overall I have been very happy with the way IBB have tried to meet my requests. If RTP traffic is prioritised then it will be perfectly suited to use VoIP.


Now here's hoping that I can get a LOS to their mast! lol!


If all is well and the IBB crew that are coming on wednesday are able to get LOS then I will be buying the rest of my SIP phones and should be fully up and running by the week after next!

Overlander

I recently got digiweb Lite (which has 1MB down 256 up and no static IP address.)
I plugged in my Budgetone Grandstream phone and tried it with STUN on and off, but after a few minutes it I cannot call to it because the network loses track of it.
I tried setting the Keep-alive value from the default 20 seconds to 10 (the minimum). But still the phone doesn't get calls always.
I tried WINSTUN to the Freespeech stun server (my voip provider) and it gave the following result:
> Nat with Independend Mapping and Port Dependent Filter - VoIP will work with
> STUN
> Preserves port number
> Does not supports hairpin of media
> Public IP address: 83.147.xxx.xxx (xxx not disclosed on bulletin board)

Also I can't activate port forwarding to 5060 on my router because I don't have access to the admin settings on the Thompson Cable modem.


Has anyone overcome this problem? Have Digiweb successfully stopped me from using a VoIP phone? (At 26cents per min to call a mobile I hope not!)
Thanks.