This has been doing my head in and I need some assistance.
I used to operate Asterisk via the FreePBX GUI, everything worked well. I now want to run it via the command line as a learning experience so on a clean OS (tried Ubuntu and now CentOS) I have an asterisk 10.1.2 (but I get the same issues on 1.8) install and am adding my config to the top of the appropriate sample configs. My testing shows internal calls and calls outbound over the trunks work ok but there are no registrations with the provider therefore no inbound calls.
I have two trunks from the same provider and both have the same issue.
sip.conf
---------------------------
[iTalkDUN]
disallow=all
username=6439746XYZ
fromuser=6439746XYZ
type=friend
secret=MYSECRET
qualify=yes
nat=no
insecure=port,invite
host=dud.italk.co.nz
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
register => 6439746XYZ:MYSECRET@dud.italk.co.nz/6439746XYZ
With sip set debug on I can see there is definitely an issue, but I can configure X-lite up with the same details and it works. Hence I don't think it is a service provider issue.
Here are the debugs. Note this is hosted on a VPS which I believe doesn't have NAT. I have however set NAT=yes and the same occurs. This was working well under FreePBX on the same site and IP address so am stumped why it fails now.
Reliably Transmitting (no NAT) to 203.184.16.2:5060:
OPTIONS sip:dud.italk.co.nz SIP/2.0
Via: SIP/2.0/UDP 95.154.207.166:5060;branch=z9hG4bK60316204
Max-Forwards: 70
From: "asterisk" <sip:6439746XYZ@95.154.207.166>;tag=as78269ddd
To: <sip:dud.italk.co.nz>
Contact: <sip:6439746XYZ@95.154.207.166:5060>
Call-ID: 383890ab26e6cdaa6dc49c442a222103@95.154.207.166:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Tue, 21 Feb 2012 22:15:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:203.184.16.2:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 95.154.207.166:5060;branch=z9hG4bK60316204;received=95.154.207.166
From: "asterisk" <sip:6439746XYZ@95.154.207.166>;tag=as78269ddd
To: <sip:dud.italk.co.nz>;tag=as42e8883c
Call-ID: 383890ab26e6cdaa6dc49c442a222103@95.154.207.166:5060
CSeq: 102 OPTIONS
User-Agent: italk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
Any help is appreciated
Core: CS1000 High Availability rls 6.0
Apps: Contact Centre 7.0, CallPilot, Telephony Manager, WLAN IP Telephony Manager
IP Devices: 1120E, 1140E, 1150E, 2033, 6120, 6140
I used to operate Asterisk via the FreePBX GUI, everything worked well. I now want to run it via the command line as a learning experience so on a clean OS (tried Ubuntu and now CentOS) I have an asterisk 10.1.2 (but I get the same issues on 1.8) install and am adding my config to the top of the appropriate sample configs. My testing shows internal calls and calls outbound over the trunks work ok but there are no registrations with the provider therefore no inbound calls.
I have two trunks from the same provider and both have the same issue.
sip.conf
---------------------------
[iTalkDUN]
disallow=all
username=6439746XYZ
fromuser=6439746XYZ
type=friend
secret=MYSECRET
qualify=yes
nat=no
insecure=port,invite
host=dud.italk.co.nz
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
register => 6439746XYZ:MYSECRET@dud.italk.co.nz/6439746XYZ
With sip set debug on I can see there is definitely an issue, but I can configure X-lite up with the same details and it works. Hence I don't think it is a service provider issue.
Here are the debugs. Note this is hosted on a VPS which I believe doesn't have NAT. I have however set NAT=yes and the same occurs. This was working well under FreePBX on the same site and IP address so am stumped why it fails now.
Reliably Transmitting (no NAT) to 203.184.16.2:5060:
OPTIONS sip:dud.italk.co.nz SIP/2.0
Via: SIP/2.0/UDP 95.154.207.166:5060;branch=z9hG4bK60316204
Max-Forwards: 70
From: "asterisk" <sip:6439746XYZ@95.154.207.166>;tag=as78269ddd
To: <sip:dud.italk.co.nz>
Contact: <sip:6439746XYZ@95.154.207.166:5060>
Call-ID: 383890ab26e6cdaa6dc49c442a222103@95.154.207.166:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Tue, 21 Feb 2012 22:15:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:203.184.16.2:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 95.154.207.166:5060;branch=z9hG4bK60316204;received=95.154.207.166
From: "asterisk" <sip:6439746XYZ@95.154.207.166>;tag=as78269ddd
To: <sip:dud.italk.co.nz>;tag=as42e8883c
Call-ID: 383890ab26e6cdaa6dc49c442a222103@95.154.207.166:5060
CSeq: 102 OPTIONS
User-Agent: italk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
Any help is appreciated
Core: CS1000 High Availability rls 6.0
Apps: Contact Centre 7.0, CallPilot, Telephony Manager, WLAN IP Telephony Manager
IP Devices: 1120E, 1140E, 1150E, 2033, 6120, 6140