Can't receive incoming calls from sip trunk

General Help
#1

I just setup AsteriskNow on a system for the first time. Once I had it installed I setup a sip trunk and two extensions. I have an any DID/CID inbound route setup to go to extention 101. I am able to call between the extensions and call out on the sip-trunk. I can’t however receive calls in on the trunk. When I dial the DID it is just dead silence then after about 15 sec it hangs up on me. I thought maybe it was my provider so I setup the trunk on a softphone and was able to recieve calls. I will list the setup of the sip-trunk below if anyone has any insight I would greatly appriciate it. Thanks

Sip-Trunk Settings

*General Settings

The only thing I set in this section was maximum channels=2

*Outgoing Settings

Trunk Name=VoipVoip

Peer Details

username=(VOIPVOIP.COM NUMBER)
type=peer
secret=(passcode)
nat=auto
insecure=very
host=69.90.209.57
fromuser=5557745602
fromdomain=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=g729&ilbc&ulaw&alaw

*Incoming Settings

USER Context=from-trunk

USER Details

username=(VOIPVOIP.COM NUMBER)
type=user
secret=(passcode)
nat=auto
insecure=very
host=69.90.209.57
fromdomain=69.90.209.57
dtmfmode=rfc2833
context=from-trunk
disallow=all
allow=g729&ulaw&alaw&ilbc

*Registration

Register String=(VOIPVOIP.COM NUMBER):(passcode)@69.90.209.57/(VOIPVOIP.COM NUMBER)

For incoming DID calls add srvlookup=no to SIP_GENERAL_CUSTOM_CONF

These are the settings that the provider listed as the proper setup for asterisk.

#2

Do you see the call comming in on the CLI? If not then the trunk isn’t properly registering with your provider. If you do see the call comming in post the call processing here. I’m assuming that where you show (passcode) you have your correct password entered. If you are behind a firewall I’d recommend changing nat to yes.

run asterisk -rvvvvvv to get to the CLI from there you can do help to see other possible commands including debug commands that might help troubleshoot the issue.

#3

I have the same problem. The call is comming and I get this:

<— SIP read from remote.server.ip:5060 —>
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: [email protected]
Contact: sip:remote.server.ip:5060
Content-Type: application/sdp
CSeq: 74336149 INVITE
From: “0727966522” sip:[email protected];user=phone;tag=11546-YZ-047333ef-430d22b17
Max-Forwards: 31
P-Preferred-Identity: sip:[email protected];user=phone
To: sip:[email protected];user=phone
User-Agent: Cirpack/v4.41f (gw_sip)
Via: SIP/2.0/UDP remote.server.ip:5060;branch=z9hG4bK-2FCD-805B9
Content-Length: 369

v=0
o=cp10 125976170740 125976170740 IN IP4 212.146.99.9
s=SIP Call
c=IN IP4 212.146.99.9
t=0 0
m=audio 30738 RTP/AVP 4 8 18 125 101
b=AS:64
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (13 headers 17 lines) —
Sending to remote.server.ip : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘GTSAMM’

<— Reliably Transmitting (no NAT) to remote.server.ip:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP remote.server.ip:5060;branch=z9hG4bK-2FCD-805B9;received=remote.server.ip
From: “0727966522” sip:[email protected];user=phone;tag=11546-YZ-047333ef-430d22b17
To: sip:[email protected];user=phone;tag=as24dfe3b1
Call-ID: [email protected]
CSeq: 74336149 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="69632056"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
localhost*CLI>
<— SIP read from remote.server.ip:5060 —>
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email protected]
Contact: sip:remote.server.ip:5060
CSeq: 74336149 ACK
From: “0727966522” sip:[email protected];user=phone;tag=11546-YZ-047333ef-430d22b17
Max-Forwards: 31
To: sip:[email protected];user=phone;tag=as24dfe3b1
User-Agent: Cirpack/v4.41f (gw_sip)
Via: SIP/2.0/UDP remote.server.ip:5060;branch=z9hG4bK-2FCD-805B9
Content-Length: 0

localhostCLI>
<------------->
— (10 headers 0 lines) —
localhostCLI>
<— SIP read from remote.server.ip:5060 —>
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: [email protected]
Contact: sip:remote.server.ip:5060
Content-Type: application/sdp
CSeq: 74336151 INVITE
From: “0727966522” sip:[email protected];user=phone;tag=11546-YZ-047333ef-430d22b17
Max-Forwards: 31
P-Preferred-Identity: sip:[email protected];user=phone
Proxy-Authorization: Digest username=“anonymous”,realm=“asterisk”,nonce=“69632056”,uri=“sip:[email protected]:5060”,response=“b0375de021189e26d45f6d39223977a9”,algorithm=MD5,opaque=""
To: sip:[email protected];user=phone
User-Agent: Cirpack/v4.41f (gw_sip)
Via: SIP/2.0/UDP remote.server.ip:5060;branch=z9hG4bK-5308-805BD
Content-Length: 369

v=0
o=cp10 125976170742 125976170742 IN IP4 212.146.99.9
s=SIP Call
c=IN IP4 212.146.99.9
t=0 0
m=audio 31448 RTP/AVP 4 8 18 125 101
b=AS:64
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (14 headers 17 lines) —
Sending to remote.server.ip : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘GTSAMM’

<— Reliably Transmitting (no NAT) to remote.server.ip:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP remote.server.ip:5060;branch=z9hG4bK-5308-805BD;received=remote.server.ip
From: “0727966522” sip:[email protected];user=phone;tag=11546-YZ-047333ef-430d22b17
To: sip:[email protected];user=phone;tag=as24dfe3b1
Call-ID: [email protected]
CSeq: 74336151 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
localhost*CLI>
<— SIP read from remote.server.ip:5060 —>
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email protected]
Contact: sip:remote.server.ip:5060
CSeq: 74336151 ACK
From: “0727966522” sip:[email protected];user=phone;tag=11546-YZ-047333ef-430d22b17
Max-Forwards: 31
Proxy-Authorization: Digest username=“anonymous”,realm=“asterisk”,nonce=“69632056”,uri=“sip:[email protected]:5060”,response=“b0375de021189e26d45f6d39223977a9”,algorithm=MD5,opaque=""
To: sip:[email protected];user=phone;tag=as24dfe3b1
User-Agent: Cirpack/v4.41f (gw_sip)
Via: SIP/2.0/UDP remote.server.ip:5060;branch=z9hG4bK-5308-805BD
Content-Length: 0

any ideea?

Thanks

#4

Your PEER settings are missing a “context=”. Yes, PEER settings.

username=(VOIPVOIP.COM NUMBER)
type=peer
secret=(passcode)
nat=auto

context=from-trunk

insecure=very
host=69.90.209.57
fromuser=5557745602
fromdomain=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=g729&ilbc&ulaw&alaw

NAT setting should be nat=yes, if you are, in fact, operating in a NAT scenario.

/S

#5