[SOLVED] incoming not working?


#1

Hi,
I’ve setup Asterisk and have it working sucessfully with outgoing calls (I’m using Telepacket as my provider). Unfortunately, incoming calls are going straight to my voicemail on Telepacket. I ran the CLI and turned on sip debugging. When I called, I got the following. Help?

…dave

asterisk1*CLI>
<-- SIP read from 67.43.159.38:5060:
INVITE sip:s@24.95.62.160 SIP/2.0
Record-Route: sip:67.43.159.38;ftag=110FD308-2653;lr=on
Record-Route: sip:207.71.126.132;ftag=110FD308-2653;lr=on
Record-Route: sip:207.71.120.251;ftag=110FD308-2653;lr=on
Via: SIP/2.0/UDP 67.43.159.38;branch=z9hG4bK8a08.3dff39a99bb8c55cb05fab2739f0c4e7.0
Via: SIP/2.0/UDP 207.71.126.132;branch=z9hG4bK8a08.5d71a752.0
Via: SIP/2.0/UDP 207.71.120.251;branch=z9hG4bK8a08.eeaf2c61.0
Via: SIP/2.0/UDP 67.43.155.130:5060
From: sip:16144752185@67.43.155.130;tag=110FD308-2653
To: sip:16143408887@207.71.120.251
Call-ID: E6B4C03-5C8411DA-B3E5DE0F-241EC949@67.43.155.130
Supported: timer,100rel
Min-SE: 1800
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 3
Remote-Party-ID: sip:+16144752185@67.43.155.130;party=calling;screen=yes;privacy=off
Contact: sip:16144752185@67.43.155.130:5060
Expires: 600
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 377

v=0
o=CiscoSystemsSIP-GW-UserAgent 2120 5824 IN IP4 67.43.155.130
s=SIP Call
c=IN IP4 67.43.155.130
t=0 0
m=audio 18918 RTP/AVP 18 0 3 100 101
c=IN IP4 67.43.155.130
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:active

— (22 headers 16 lines)—
Using INVITE request as basis request - E6B4C03-5C8411DA-B3E5DE0F-241EC949@67.43.155.130
Sending to 67.43.159.38 : 5060 (non-NAT)
Found peer 'telepacket-out’
Reliably Transmitting (no NAT) to 67.43.159.38:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 67.43.159.38;branch=z9hG4bK8a08.3dff39a99bb8c55cb05fab2739f0c4e7.0;received=67.43.159.38
Via: SIP/2.0/UDP 207.71.126.132;branch=z9hG4bK8a08.5d71a752.0
Via: SIP/2.0/UDP 207.71.120.251;branch=z9hG4bK8a08.eeaf2c61.0
Via: SIP/2.0/UDP 67.43.155.130:5060
From: sip:16144752185@67.43.155.130;tag=110FD308-2653
To: sip:16143408887@207.71.120.251;tag=as660251e7
Call-ID: E6B4C03-5C8411DA-B3E5DE0F-241EC949@67.43.155.130
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:s@24.95.62.160
Proxy-Authenticate: Digest realm=“asterisk”, nonce="3039b13b"
Content-Length: 0


Scheduling destruction of call ‘E6B4C03-5C8411DA-B3E5DE0F-241EC949@67.43.155.130’ in 15000 ms
asterisk1*CLI>
<-- SIP read from 67.43.159.38:5060:
ACK sip:s@24.95.62.160 SIP/2.0
Via: SIP/2.0/UDP 67.43.159.38;branch=z9hG4bK8a08.3dff39a99bb8c55cb05fab2739f0c4e7.0
From: sip:16144752185@67.43.155.130;tag=110FD308-2653
Call-ID: E6B4C03-5C8411DA-B3E5DE0F-241EC949@67.43.155.130
To: sip:16143408887@207.71.120.251;tag=as660251e7
CSeq: 101 ACK
Content-Length: 0

— (7 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 67.43.159.38:5060:
OPTIONS sip:s@24.95.62.160 SIP/2.0
Via: SIP/2.0/UDP 67.43.159.38:5060;branch=0
From: sip:pinger@telepacket.com;tag=85f7c051
To: sip:s@24.95.62.160
Call-ID: 7d599364-24940686-87ee2@67.43.159.38
CSeq: 1 OPTIONS
Content-Length: 0

— (7 headers 0 lines)—
Looking for s in from-sip-external (domain 24.95.62.160)
Transmitting (no NAT) to 67.43.159.38:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.43.159.38:5060;branch=0;received=67.43.159.38
From: sip:pinger@telepacket.com;tag=85f7c051
To: sip:s@24.95.62.160;tag=as68e9218a
Call-ID: 7d599364-24940686-87ee2@67.43.159.38
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:24.95.62.160
Accept: application/sdp
Content-Length: 0


Destroying call '7d599364-24940686-87ee2@67.43.159.38’
asterisk1CLI>
Destroying call 'E6B4C03-5C8411DA-B3E5DE0F-241EC949@67.43.155.130’
asterisk1
CLI>
[/code]


#2

Do you have a 16143408887 extension in extensions.conf? Is it in the correct context?


#3

Please see my reply to in this thread, forums.digium.com/viewtopic.php?t=2597, it should help you.

Regards.

Marco Bruni


#4

Hi,
I got this fixed (before I saw your reply) by following the instructions at:

samyantoun.50webs.com/asterisk/a … allpacket/

Thanks anyway. :smile:

…dave