Can't receive incoming calls

bartigas · November 24, 2009, 11:38pm

Hello,

I just setup a FreePBX in the datacenter and am using X-Lite to test everything. I’m having a difficult time getting FreePBX to pass through an incoming call. I can make outgoing calls without a problem. When I try to call in to the system I get “The number you have dialed is not in service, please check the number and try again.” Here are the details:

For security reasons I blanked out the last two octets of the ip addresses, the last 4 numbers of the phone number and the passwords.

h = ip address digit for FreePBX
i = ip address digit for SER Proxy server
j = ip address digit for provider’s ip
n = called phone number (in)
p = calling phone number (out)
x = password character

Trunk Settings:

Outbound Caller ID:561935nnnn

Trunk Name: test

Peer Details:
host=208.115.iii.iii
username=1561935nnnn
secret=xxxxxx
type=peer
insecure=very

User Context: 1561935nnnn

User Details:
secret=xxxxxx
type=peer
context=from-trunk
permit=208.115.iii.iii

Register String: 1561935nnnn:[email protected]

CLI Output

<— SIP read from 208.115.III.III:5060 —>
INVITE sip:[email protected] SIP/2.0
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKef4c.0be495b3.0
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK5344ecfc;rport=5060
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected]
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: TSG_GLOBAL_COBRA
Max-Forwards: 16
Remote-Party-ID: sip:[email protected];privacy=off;screen=no
Date: Tue, 24 Nov 2009 23:16:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
X-ANI-II: N/A
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 31343 31343 IN IP4 209.247.22.141
s=session
c=IN IP4 209.247.22.141
t=0 0
m=audio 60022 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
— (18 headers 14 lines) —
Sending to 208.115.III.III : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer 'test1’
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.247.22.141:60022
Looking for s in from-sip-external (domain 208.115.HHH.HHH)
list_route: hop: sip:208.115.III.III;ftag=as7db4c12a;lr=on

<— Transmitting (no NAT) to 208.115.III.III:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKef4c.0be495b3.0;received=208.115.III.III
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK5344ecfc;rport=5060
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [s@from-sip-external:1] GotoIf(“SIP/test1-00000038”, “0?from-trunk||1”) in new stack
– Executing [s@from-sip-external:2] Set(“SIP/test1-00000038”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2009-11-24 23:50:14 UTC.
– Executing [s@from-sip-external:3] Answer(“SIP/test1-00000038”, “”) in new stack
Audio is at 208.115.HHH.HHH port 16422
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
freepbx*CLI>
<— Reliably Transmitting (no NAT) to 208.115.III.III:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKef4c.0be495b3.0;received=208.115.III.III
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK5344ecfc;rport=5060
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2396 2396 IN IP4 208.115.HHH.HHH
s=session
c=IN IP4 208.115.HHH.HHH
t=0 0
m=audio 16422 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Executing [s@from-sip-external:4] Wait(“SIP/test1-00000038”, “2”) in new stack
freepbx*CLI>
<— SIP read from 208.115.III.III:5060 —>
ACK sip:[email protected] SIP/2.0
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
Via: SIP/2.0/UDP 208.115.III.III;branch=0
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK0ce5f15c;rport=5060
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: TSG_GLOBAL_COBRA
Max-Forwards: 16
Remote-Party-ID: sip:[email protected];privacy=off;screen=no
Content-Length: 0

<------------->
— (13 headers 0 lines) —
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 208.115.III.III:5060:
REGISTER sip:208.115.III.III SIP/2.0
Via: SIP/2.0/UDP 208.115.HHH.HHH:5060;branch=z9hG4bK5f44933f;rport
From: sip:[email protected];tag=as11990e7a
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 113 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“1561935NNNN”, realm=“208.115.III.III”, algorithm=MD5, uri=“sip:208.115.III.III”, nonce=“4b0c9c07044289f6b80ed101108139cb8e4b02fc”, response="3c33c6fe8851a79e24666866e121997d"
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0

freepbx*CLI>
<— SIP read from 208.115.III.III:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.115.HHH.HHH:5060;branch=z9hG4bK5f44933f;rport=5060
From: sip:[email protected];tag=as11990e7a
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 113 REGISTER
Server: Sip EXpress router (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 208.115.III.III:5060 “Noisy feedback tells: pid=1231 req_src_ip=208.115.HHH.HHH req_src_port=5060 in_uri=sip:208.115.III.III out_uri=sip:208.115.III.III via_cnt==1”

<------------->
— (9 headers 0 lines) —

<— SIP read from 208.115.III.III:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.115.HHH.HHH:5060;branch=z9hG4bK5f44933f;rport=5060
From: sip:[email protected];tag=as11990e7a
To: sip:[email protected];tag=d013a8929d2f93884fadca19e2cdf071.0fd6
Call-ID: [email protected]
CSeq: 113 REGISTER
Contact: sip:[email protected];expires=120
Server: Sip EXpress router (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 208.115.III.III:5060 “Noisy feedback tells: pid=1231 req_src_ip=208.115.HHH.HHH req_src_port=5060 in_uri=sip:208.115.III.III out_uri=sip:208.115.III.III via_cnt==1”

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
– Executing [s@from-sip-external:5] Playback(“SIP/test1-00000038”, “ss-noservice”) in new stack
– <SIP/test1-00000038> Playing ‘ss-noservice’ (language ‘en’)
freepbx*CLI>
<— SIP read from 208.115.III.III:5060 —>
INVITE sip:[email protected] SIP/2.0
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKff4c.68fc73a3.0
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK735853aa;rport=5060
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: TSG_GLOBAL_COBRA
Max-Forwards: 16
Remote-Party-ID: sip:[email protected];privacy=off;screen=no
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 31343 31344 IN IP4 69.25.JJJ.JJJ
s=session
c=IN IP4 69.25.JJJ.JJJ
t=0 0
m=audio 34882 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
— (16 headers 12 lines) —
Sending to 208.115.III.III : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 69.25.JJJ.JJJ:34882

<— Transmitting (no NAT) to 208.115.III.III:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKff4c.68fc73a3.0;received=208.115.III.III
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK735853aa;rport=5060
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
Audio is at 208.115.HHH.HHH port 16422
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 208.115.III.III:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKff4c.68fc73a3.0;received=208.115.III.III
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK735853aa;rport=5060
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2396 2397 IN IP4 208.115.HHH.HHH
s=session
c=IN IP4 208.115.HHH.HHH
t=0 0
m=audio 16422 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
freepbx*CLI>
<— SIP read from 208.115.III.III:5060 —>
ACK sip:[email protected] SIP/2.0
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
Via: SIP/2.0/UDP 208.115.III.III;branch=0
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK309adca6;rport=5060
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: TSG_GLOBAL_COBRA
Max-Forwards: 16
Remote-Party-ID: sip:[email protected];privacy=off;screen=no
Content-Length: 0

<------------->
— (13 headers 0 lines) —
freepbx*CLI>
<— SIP read from 208.115.III.III:5060 —>
BYE sip:[email protected] SIP/2.0
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKcf4c.a3539d85.0
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK4d3efe30;rport=5060
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: TSG_GLOBAL_COBRA
Max-Forwards: 16
Remote-Party-ID: sip:[email protected];privacy=off;screen=no
X-TSG_GLOBAL-HangupCause: Normal Clearing
X-TSG_GLOBAL-HangupCauseCode: 16
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 208.115.III.III : 5060 (no NAT)

<— Transmitting (no NAT) to 208.115.III.III:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.115.III.III;branch=z9hG4bKcf4c.a3539d85.0;received=208.115.III.III
Via: SIP/2.0/UDP 69.25.JJJ.JJJ:5060;branch=z9hG4bK4d3efe30;rport=5060
Record-Route: sip:208.115.III.III;ftag=as7db4c12a;lr=on
From: sip:[email protected];tag=as7db4c12a
To: sip:[email protected];tag=as004063a2
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
== Spawn extension (from-sip-external, s, 5) exited non-zero on ‘SIP/test1-00000038’
– Executing [h@from-sip-external:1] NoOp(“SIP/test1-00000038”, “Hangup”) in new stack
– Executing [h@from-sip-external:2] Set(“SIP/test1-00000038”, “DID=s”) in new stack
– Executing [h@from-sip-external:3] Goto(“SIP/test1-00000038”, “s|1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/test1-00000038”, “0?from-trunk|s|1”) in new stack
– Executing [s@from-sip-external:2] Set(“SIP/test1-00000038”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2009-11-24 23:50:19 UTC.
– Executing [s@from-sip-external:3] Answer(“SIP/test1-00000038”, “”) in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/test1-00000038’
Really destroying SIP dialog ‘[email protected]’ Method: BYE
freepbx*CLI>
<— SIP read from 216.199.214.66:58395 —>

What baffles me is the “INVITE sip:[email protected] SIP/2.0” on my other commercial pbx it shows the phone number (ie. [email protected]), why doesn’t this show an s?

Any help would be greatly appreciated, I’m begining to devlop a flat spot on my forehead from banging it against the wall.

Thank in advance,

Brian

Cliffster · November 25, 2009, 3:27pm

I don’t understand it very well, but I had to change my context from context=from-pstn to context=from-zaptel

Hope that helps,

Cliffster

bartigas · November 25, 2009, 3:50pm

I tried that and it still didn’t work. Same result as before.

Cliffster · November 25, 2009, 4:16pm

I always bring up a PuTTY session to it and look at the logs using tail:
tail -f --lines=400 /var/log/asterisk/full

What kind of trunk are you using and how do you have inbound routes setup?

You did good getting the outbound calls to work. Dial plans and patterns take a bit of real thinking.

Cliffster · November 25, 2009, 4:31pm

Mine is a Zap trunk but

Check allow unauthenticated SIP calls. Under the General Settings tab

Cliffster · November 25, 2009, 4:33pm

oop

Cliffster · November 25, 2009, 4:33pm

Allow Anonymous Inbound SIP Calls

bartigas · November 27, 2009, 3:10pm

Hey Cliffster thanks so much for all of your assistance. It actually turned out to be a problem with the sip register string.

I had: Register String: 1561935nnnn:[email protected]

It needed to be: Register String: 1561935nnnn:[email protected]/1561935nnnn

I also needed to turn on Allow Anonymous Inbound SIP Calls.

We’re using termination and origination from our provider going straight into a SER server which basically turns it into trunks and shoots it out to a number of servers and devices.

Thanks again for your help, it was much appreciated.

Brian

SkykingOH · November 27, 2009, 6:35pm

Why don’t you setup a peer to the OpenSER box?

host=IP of SER
insecure=very
type=friend
disallow=all
allow=ulaw

This config will authenticate the SIP packets to the peer bases on the IP address only.