[Asterisk-Users] iconnect

[Shoval Tomer]

Hi,

I was able to connect asterisk to iconnect's service.

It took me almost two hours, but it's because I was having NAT trouble.

I finally discovered that you can set the iconnect host to
natrealy.deltathree.com to make it work.

 

(for those of you who, like me, don't have the time to search the
archive I'll provide a working sample in a minute)

 

My problem was sound quality. It was horrible. It took as much as five
seconds to bridge iconnect to my sip soft phone, and then the voice
recording on the other side (I called our branch office, and everyone
has gone for the night) was lousy. Stuttering like and broken.

 

Here's my setup.

 

A PII 400 with 256MB of RAM for Asterisk.

x-lite running on a PIV 1.8Ghz with 512MB of RAM (my thinkpad)

 

now get the network config

 

I was at home (ADSL line - 0.5G download, 64Kbps upload)

I connected to HQ (where asterisk is - ADSL line, 2.5G download, 128Kbps
upload)

The connection was made via VPN (cisco vpn client on my side and Cisco
VPN concentrator on the other side)

 

The Asterisk server is behind NAT, and so am I at home, but that
shouldn't count for performance, IMHO as AFAIK (cool, first time to use
those (-: )

 

I might add the HQ is in Israel, and as far as I can tell, iconnect
servers at deltathree are on the other side of the globe.

 

As for bandwidth consumption at the time -

I closed EVERYTHING on my machine.

HQ is empty (it's almost 2:00 AM here)

The branch office is also empty now (I was trying to call a pstn line
there via iconnect, that's not relevant to Asterisk or VOIP, it's just
that they, too, connect to HQ from there via VPN, but as I've said - it
was empty - I got the answering machine there)

 

Where should I look?

Is this SIP related?

Is this asterisk related (server hardware - I think ram is pretty much
maxed out)

 

I was intending to get it to work and show it to my boss tomorrow, which
would pretty much mean the asterisk will be doing some production time
starting tomorrow afternoon, but...

 

Just for comparison - I have called the same phone number via iconnect
without asterisk in the middle using their pc2phone sip client, about a
minute after I tried with my sip client and asterisk and performance was
about three times better (clear voice, no background noise and a much
much shorter delay)

 

This performance was adequate for calling through the net using ms
messenger for kicks, not for a production system you intend to use to
provide customer support.

 

And ideas will be gladly accepted.

 

Now for the sample:

 

Sip.conf:

 

[general]

port=5060

bindaddr=0.0.0.0

context=from-sip

callerid=NO CallID

 

 

[iconnect]                      ;sip for iconnect

type=friend

username=12345678

secret=1234

host=natrelay.deltathree.com

 

[1000]                           ;my sip extension

type=friend

secret=1234

iauth=md5

nat=yes

host=dynamic

reinvite=no

canreinvite=no

qualify=1000

dtmfmode=rfc2883

callerid="Tomer Shoval" <1000>

disallow=all

allow=gsm

context=default

 

Extensions.conf

 

[general]

static=yes

writeprotect=yes

 

[globals]

 

 

[default]

exten => _91.,1,Dial(SIP/${EXTEN:1}@iconnect,70)

exten => _91.,2,Hangup

 

As you can see, very minimal, for this kind of usage (outbound iconnect)
only and from one sip phone only.

These are the actual configuration files, with usernames and passwords
garbled.

 

 

Thanks.

 

 

Shoval

 

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