[Asterisk-Users] Iconnecthere

[Andrew Joakimsen]

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Armand A.
Verstappen
Sent: Sunday, August 10, 2003 5:29 PM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] Iconnecthere

Hi Andrew,

On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote:
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Armand A.
> Verstappen
> Sent: Sunday, August 10, 2003 4:57 AM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Iconnecthere
> 
> On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote:
> > On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote:
> > > Does anyone have Asterisk working with Iconnect here for incoming
> > and/or
> > > outgoing calls? 
> > 
> > have a look at:
> > 
> >
>
http://www.loligo.com/asterisk/example-configs.2003-04-24/extensions.con
> > f
> > 
> > there's a section in there dealing with Iconnect
> 
> > That helps a lot. But now I get this message when I try to dial any
> > number
> > 
> > NOTICE[5126]: File pbx.c, Line 1089 (pbx_extension_helper): Cannot
> find
> > extension context 'default'
> 
> You get this notice doing what? dialing in, dialing out? At any rate,
it
> looks like you've halfway implemented the examples, sip.conf having
> context=default, but no context [default] in extensions.conf.
> 
> When I try to dial out, but there is a [default] section in my
> extensions.conf

Hmm... okay. We're going to need a little more context here. What kind
of device/software are you calling from? sip / h323 / zap / quicknet /
... ? What's the related config setup like (so sip.conf
h323.conf/oh323.conf zap.conf phone.conf ...), and what are your
extensions set up like (extensions.conf).

wkr


I seem to have my configuration working except for outgoing and incoming
calls for the rest of the world. For now I am concerned more about
outgoing calls than anything else. Whenever I try to make an outgoing
call I get these messages from the sip debug in the console

s=session
c=IN IP4 64.36.104.203
t=0 0
m=audio 6620 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (no NAT) to 4.42.235.170:5060
    -- Called 13057400221 at packet8.net
Sip read:
SIP/2.0 404 Not Found 4
Via: SIP/2.0/UDP 64.36.104.203:5060;branch=z9hG4bK37d8c90a
From: "asterisk" <sip:asterisk at 64.36.104.203>;tag=as220b2c68
To: <sip:13057400221 at 4.42.235.170>;tag=1m6lkhivci11cjdooja30ex45
Call-ID: 72a3aaff62d64ab064938da36f1dcdb9 at 64.36.104.203
CSeq: 102 INVITE
Content-Length: 0


Notice in particular the From line. Now notice a working session from
eStara softphone:


v=0
o=eStara 22079953 22079953 IN IP4 64.36.104.202
s=eStara
c=IN IP4 64.36.104.202
t=0 0
m=audio 8014 RTP/AVP 0 4 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.36.104.202:5060
From: Anonymous <sip:17862324057 at packet8.net>;tag=1d436a9
To: <sip:13057400221 at packet8.net>
Call-ID: ed3d1f22-3805-4117-a5ed-8df16f7cd5f9 at 64.36.104.202
CSeq: 22079953 INVITE
Content-Length: 0


Notice how the from is different, my SIP service will not accept calls
unless the proper from name is configured, how can I configure this?
Here are the relevant sections from my sip.conf file

[general]
port = 5060			; Port to bind to
context = from-sip		; Default for incoming calls
maxexpirey=13600		; Max length of incoming registration we
allow
defaultexpirey=3600		; Default length of incoming/outoing
registration
register => 17862324057:xxxxxxxxxxx at packet8.net/5500

[packet8.net]
type=friend
username=17862324057
secret=xxxxxxxxxxx
host=packet8.net
context=demo



Thanks again for all the help you have provided me with.