Inbound Route . Not Working

Asterisk Asterisk Support
1

The inboound route dosen’t work at all. I’ve tried diffrent VoIP providers, but still the same problme.

The inbound on the PSTN line, ZAP/g1 trunk works fine.
Inbound on any SIP, no chance. Since the outbound works I don’t think that there is a register problem with the VoIP provider.

Sip Registry

Name/username Host Dyn Nat ACL Port Status
GIZMO/17476839831 198.65.166.131 N 5060 Unmonitored
FreeDigits/5637733205 67.55.159.160 N 5060 Unmonitored
1010/1010 (Unspecified) D 0 Unmonitored
1001/1001 192.168.1.100 D 13596 Unmonitored
4 sip peers [4 online , 0 offline]
Verbosity is at least 2
Core debug is at least 1
== Manager ‘admin’ logged on from 127.0.0.1

Active Channel(s)

Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
198.65.166.131 1747683983 1c04d93865b 00103/00000 unkn No
67.55.159.160 5637733205 3ef789d362f 00103/00000 unkn No
2 active SIP channels
Verbosity is at least 2
Core debug is at least 1

THE DEBUG message is :

<-- SIP read from 67.55.159.160:5060:
INVITE sip:s@67.175.188.139:5060 SIP/2.0
Record-Route: sip:67.55.159.160;ftag=as2e3bdcad;lr=on
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKb679.8c6096e7.0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK066cda1c;rport =5060
From: “3124795661” sip:3124795661@67.55.159.156;tag=as2e3bdcad
To: sip:5637733205@freedigits.net
Contact: sip:3124795661@67.55.159.156:5060
Call-ID: 782d0a884c2a2a733c48d65a1060fea0@67.55.159.156
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 16
Date: Wed, 09 Aug 2006 23:25:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY
Content-Type: application/sdp
Content-Length: 441

v=0
o=root 12519 12519 IN IP4 67.55.159.156
s=session
c=IN IP4 67.55.159.156
t=0 0
m=audio 15318 RTP/AVP 0 8 3 18 111 5 10 7 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

— (15 headers 19 lines)—
Using INVITE request as basis request - 782d0a884c2a2a733c48d65a 1060fea0@67.55.159.156
Sending to 67.55.159.160 : 5060 (non-NAT)
Found peer 'FreeDigits’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 10
Found RTP audio format 7
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 67.55.159.156:15318
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format G729
Found description format G726-32
Found description format DVI4
Found description format L16
Found description format LPC
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x5fe (gsm|ulaw |alaw|g726|adpcm|slin|lpc10|g729|ilbc)/video=0x0 (nothing), comb ined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 ( telephone-event), combined - 0x1 (telephone-event)
Looking for s in from-sip-external (domain 67.175.188.139)
list_route: hop: sip:67.55.159.160;ftag=as2e3bdcad;lr=on
Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKb679.8c6096e7.0;rec eived=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK066cda1c;rport =5060
From: “3124795661” sip:3124795661@67.55.159.156;tag=as2e3bdcad
To: sip:5637733205@freedigits.net
Call-ID: 782d0a884c2a2a733c48d65a1060fea0@67.55.159.156
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY
Contact: sip:s@192.168.1.101
Content-Length: 0

We’re at 192.168.1.101 port 16206
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKb679.8c6096e7.0;rec eived=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK066cda1c;rport =5060
Record-Route: sip:67.55.159.160;ftag=as2e3bdcad;lr=on
From: “3124795661” sip:3124795661@67.55.159.156;tag=as2e3bdcad
To: sip:5637733205@freedigits.net;tag=as2f1eb7bc
Call-ID: 782d0a884c2a2a733c48d65a1060fea0@67.55.159.156
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY
Contact: sip:s@192.168.1.101
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 2776 2776 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 16206 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
ACK sip:s@67.175.188.139:5060 SIP/2.0
Via: SIP/2.0/UDP 67.55.159.160;branch=0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK77f5f260;rport =5060
From: “3124795661” sip:3124795661@67.55.159.156;tag=as2e3bdcad
To: sip:5637733205@freedigits.net;tag=as2f1eb7bc
Contact: sip:3124795661@67.55.159.156:5060
Call-ID: 782d0a884c2a2a733c48d65a1060fea0@67.55.159.156
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 16
Content-Length: 0

— (11 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 198.65.166.131:5060:

— (0 headers 0 lines) Nat keepalive —
asterisk1CLI>
<-- SIP read from 67.55.159.160:5060:
BYE sip:s@67.175.188.139:5060 SIP/2.0
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKc679.97a22d07.0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK188c7719;rport=5060
From: “3124795661” sip:3124795661@67.55.159.156;tag=as2e3bdcad
To: sip:5637733205@freedigits.net;tag=as2f1eb7bc
Contact: sip:3124795661@67.55.159.156:5060
Call-ID: 782d0a884c2a2a733c48d65a1060fea0@67.55.159.156
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 16
Content-Length: 0
asterisk1CLI>

— (11 headers 0 lines)—
Sending to 67.55.159.160 : 5060 (NAT)
Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKc679.97a22d07.0;received=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK188c7719;rport=5060
From: “3124795661” sip:3124795661@67.55.159.156;tag=as2e3bdcad
To: sip:5637733205@freedigits.net;tag=as2f1eb7bc
Call-ID: 782d0a884c2a2a733c48d65a1060fea0@67.55.159.156
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:s@192.168.1.101
Content-Length: 0

asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:

— (0 headers 0 lines) Nat keepalive —
Destroying call ‘782d0a884c2a2a733c48d65a1060fea0@67.55.159.156’

2

I belive that I have actually not configured how I was supposed to the Incoming Route of the trunk field.

Can someone paste in here his settings + the User context

Thank you

3

do

sip show registry

to find out if you box is registred to your voip provider. make sure you do registration like

register => sip account:sip password@sip registrar/DID

example
register => 12345:abc@sip.inphonex.com/776

in your extension.conf you should have

exten => 776,1,NoOP

4

The box is registered all right. I can call out, I can call IN, and while calling in I even see my number showing in the debug line.
The problem is that the * box dosen’t pick up the call, that’s why I thought that the problem is in :

context=custom-sip

extensions.conf

[custom-sip]

???

5

You need to go to the following link and use the extensions.conf settings I posted close to the bottom of the page; forums.digium.com/viewtopic.php? … ight=drwho