[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:[EMAIL PROTECTED],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