14 Sep
2013
14 Sep
'13
2:43 p.m.
Hi Charles, the sip trace was carried out on the asterisk server. I have
also disable the fw on both servers. I suspect it might be network issue
but I have no idea what.
I have also set up another server running asterisk 11 same problem so I
know its not asterisk 1.8. I get no errors on either sever and logs don't
show any issues. Its just an odd issue thats got me baffled.
Regards
Mick
On 14 Sep 2013 09:08, "Charles Chance" <charles.chance(a)sipcentric.com>
wrote:
Hi,
Could be something network related, or firewall? I know you stated phones
could register directly to Asterisk but if nothing is showing in Asterisk
SIP log, then the requests are never getting to it.
Was your trace performed on Kamailio server or Asterisk?
Cheers,
Charles
On 13 September 2013 17:12, imperium broadcast <
imperium.broadcast(a)gmail.com> wrote:
Hi all I have been trying to figure out where Im
going wrong with
following this guide:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the
forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the
Asterisk Console it doesn't show the register attempts. I have to use ngrep
to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when
passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf
"IP change to protect the innocent!"
[kamailio-2]
type=peer
host=1.1.1.1
fromdomain=1.1.1.1
context=outgoing
insecure=invite,port
directmedia=nonat
qualify=yes
disallow=all
allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080
REGISTER sip:172.16.0.110:5080 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
To: sip:102@172.16.0.110.
From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
CSeq: 10 REGISTER.
Call-ID: 4670378d1185f5b6-25000(a)127.0.0.1.
Content-Length: 0.
User-Agent: kamailio (3.3.1 (x86_64/linux)).
Contact: <sip:102@172.16.0.112:5060>.
Expires: 120.
.
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080
REGISTER sip:172.16.0.110:5080 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
To: sip:102@172.16.0.110.
From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
CSeq: 10 REGISTER.
Call-ID: 4670378d1185f5b6-25000(a)127.0.0.1.
Content-Length: 0.
User-Agent: kamailio (3.3.1 (x86_64/linux)).
Contact: <sip:102@172.16.0.112:5060>.
Expires: 120.
and I have this for my
Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK
asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address"
asterisk.bindport = "5080" desc "Asterisk Port"
kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif
--
Regards
Mick
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