Setting up SOE using SIP to Asterisk system

[wagtech]

Having a problem with audio between SOE Sip trunk and Asterisk PBX where audio seems to be one way then after a while the call disconnects (most likely to quiet timeout).

Sounds to me to be a Codec issue, does anyone know which codec is best on either side? Unless someone has other ideas...

Phil

 

[wagtech]

After a bit more investigation voice is going out of the SOE but not in. So may not be a codec issue!?

 

[h382]

Could it be an IP route missing?

 

[eholm]

I haven't gotten that far. How do I set up the Asterisk and the IP Office to talk via SIP?

 

[ZebrAYA]

I'm now working with AVAYA mainly! but with ocassional Panasonic works - and coincidencially got a case with similar behaviour, here it's

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Keywords: voip, ip, tda100, udp, port, firewall, nat, vpn

I’m setting up a ip connection inside a new company.
The company have two Offices.

Office A: public IP address
Office B: public IP address
Connection UDP port: 8000

Both offices have KX TDA100 PBX’s with their respective IP TRUNKS and IP EXTENSIONS cards.
They have two Panasonic IP phones.

When a call is made from a ip phone to any other phone inside the same LAN (using lan only) the call is established without flaw

When the call is made from an ip phone to any phone outside, at the other office (using internet) the call is established but half way, i.e. from the extension that’s inside the LAN one can speak but not to listen, and from the extension outside the LAN one can listen but not to speak (note: both sides transmit DTMF tones without trouble)

We’ve called the ISP to report this problem and says that his router is not blocking anything, but making a IP-map between ip public address and ip private address, this private address is 192.168.0.55 that corresponds to IP EXTENSION CARD ip address.

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[eholm]

I have solved the two way audio issue when making calls from IP Office to Asterisk. It requires an interesting trick on the Asterisk side. I'm willing to share if someone can respond (wagtech?) with how they may be getting calls from the Asterisk to the IP Office - we're almost there!

 

[wagtech]

eholm, would have to do with turning on NAT on the asterisk side?

What I just found was the asterisk side was sending its public IP to the SOE causing the SOE to reply back to the public address. Now that I have turned on NAT on the asterisk side it now sends only its LAN ip addy. It now works ok. Strange quirk I thought...

 

[eholm]

wagtech, no in my case nat settings seem to have no effect. I have the * and the IPO on the same subnet. I only one way audio (ipo sends but doesn't receive) when connecting through a sip trunk on the *, to an outside number. Calls over a zap trunk work fine, as do calls from any ipo set (digital, analog, h.323) to a phone on the *. My trick to make the sip to sip calls work is to establish a sip user on the *, and have the ipo authenticate to that user. It's wierd in that the user never authenticates on the * except when a call is in progress (the call shows the ipo sip trunk and sip sip user account on the trunk as active). I still haven't figured out how to make calls from the * ring on the IPO - any tips?

 

[wagtech]

Ditto for the setup.

My setup IPO with no NAT, * with NAT. IPO only calls the * or through it to outside world.

To receive calls I have calls route via the * ext. to the IPO

Are you able to make a call out with audio both ways from the * direct? Check on the * monitor how the call is progressing using sip debug ie: get packets both ways.

The only two things I can think of at present is NAT on the trunk of the * having NAT on and/or ports on the router assigned to the * for RTP (audio) packets.

May also require changes to codecs on the trunk but check calls direct from the * first....

Phil

 

[eholm]

It's good to know the ext is required.

Yes all calls work fine from the * outside. Two way audio is fine for any kind of call, including calls from the IP Office to an * ext.

My only problem then is figuring out how to get a call from the * to the IPO. Can you be more specific about how you "route via the * ext. to the IPO"?

 

[wagtech]

Ah, so you DO have audio both ways on a call, my problem was one way audio on a call.

On my setup I dont use registration on the * only set the ip address for the extension (the IPO's ip), in fact I have all the IPO's exts existing on the * with the same IP so you can dial any ext on the IPO through the * box. Sorta like a SCN setup.

You could set up a digital recptionist on the * to put calls through to the IPO or set up a group on the IPO instead of (or as well as) an ext. with a corresponding ext. number on the * pointing to the IPO ip addy.

Getting slightly off topic this may help for the * box-

http://dumbme.voipeye.com.au/trixbox/trixbox_without_tears.htm

Hope that helps
Phil

 

[wagtech]

Oh and I forgot to mention having an incoming call route on the IPO for each ext to map the * ext (or group) number to the IPO ext/group.

 

[eholm]

Thanks for the tips. I have done several scn's and am very familiar with that. I still can't seem to get a call to ring in on the ipo from the *.

Here's some detail:

I have ext 301 on the * and the same ext on the ipo.
I have in incoming route on the ipo sip trunk group routing anything to 301, and another routing 301 to 301, like a DID.

I also set the sip info for 301 on the ipo to 301.

I'm reaching the * (and through it) from the ipo using an 8N route.
I also set up an 8|xxx route on the *, with no success there (getting * message "all circuits are busy now").

when dialing 301 from a * phone I get two rings and a busy. ipo Monitor shows a couple of things that look like an indication of the problem:
"SipDebugInfo: Incoming Method, Parsing Failed 8"
"SipDebugInfo: SipTrunks: Cannot free Txn Key 2015"

Still stumped. Any ideas?

 

[wagtech]

It looks like the ipo is failing on i/c calls starting with 8xxx. But aren't you stripping 8 in the *?

Maybe try turning on SIP Tx and SIP Rx on Monitor to see what the two units are sending each other in the SIP packets.
Post here if you want.

 

[eholm]

Wagtech
Still Stuck. I am thinking it may be one of a couple of things.
1) Do you have more than 1 sip license on your ipo? I'm getting a congestion message when watching the *, and thinking the use of an extension and trunk is maybe requiring more than one channel license on the ipo.
2) Perhaps it has to do with the specific setup of the trunk and extension on the *. Can you list what you are using for your configuration here?


Thanks!

 

[wagtech]

1) Only one (trial at present, just ordered two licenses)
2) Extn setup is basic as below-
Ext cfg
exten => 201,1,Macro(exten-vm,novm,201)
exten => 201,hint,SIP/201
Sip cfg
[201]
username=201
type=friend
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=yes
mailbox=201@device
host=192.168.0.12
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <201>

 

[eholm]

Great, thanks! do you have the same detail for the trunk you have set up to the IPO? or are you just using this single extension?

 

[wagtech]

If you try this link and then I can email you the config details direct-

http://www.nontaurus.com/msgs/ToAdmin.asp

If you are party to this...

Phil