Help with Inbound calls

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.10.07 17:57:34 =~=~=~=~=~=~=~=~=~=~=~=
login as: root
[email protected]’s password:
Last login: Fri Oct 7 17:46:55 2011 from 10.0.0.3

e]0;root@localhost:~a[root@localhost ~]#
e[K[root@localhost ~]# asteriks e[Ke[Ke[Ksk -vr
Asterisk 1.6.2.11, Copyright © 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

e[0;37me[0mConnected to Asterisk 1.6.2.11 currently running on localhost (pid = 2967)
localhost*CLI>
e[0KVerbosity is at least 3

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:10.0.0.3:36268 —>

<------------->

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:196.36.166.174:5060 —>
INVITE sip:[email protected] SIP/2.0

Record-Route: sip:196.36.166.174;lr=on;did=a3e.957cd513

Via: SIP/2.0/UDP 196.36.166.174;branch=z9hG4bKb0d5.6fea1e94.0

Via: SIP/2.0/UDP 196.36.166.170:5060;received=196.36.166.170;branch=z9hG4bK47cde944;rport=5060

From: “+27814488644” sip:[email protected];tag=as4c387c47

To: sip:[email protected]

Contact: sip:[email protected]

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: eyeBeam release 3004w stamp 16863

Max-Forwards: 69

Date: Fri, 07 Oct 2011 15:58:18 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Content-Type: application/sdp

Content-Length: 431

A2B-Account: SIP1

v=0

o=root 17833 17833 IN IP4 196.36.166.170

s=session

c=IN IP4 196.36.166.172

t=0 0

m=audio 19040 RTP/AVP 18 3 4 111 0 8 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:3 GSM/8000

a=rtpmap:4 G723/8000

a=fmtp:4 annexa=no

a=rtpmap:111 G726-32/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

a=nortpproxy:yes

<------------->

e[Klocalhost*CLI>
e[0K— (17 headers 20 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 196.36.166.174 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘3480234650’ for ‘+27814488644’ from 196.36.166.174:5060

<— Reliably Transmitting (no NAT) to 196.36.166.174:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 196.36.166.174;branch=z9hG4bKb0d5.6fea1e94.0;received=196.36.166.174

Via: SIP/2.0/UDP 196.36.166.170:5060;received=196.36.166.170;branch=z9hG4bK47cde944;rport=5060

From: “+27814488644” sip:[email protected];tag=as4c387c47

To: sip:[email protected];tag=as07970e3c

Call-ID: [email protected]

CSeq: 102 INVITE

Server: FPBX-2.8.0alpha0(1.6.2.11)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“4766b239”

Content-Length: 0

<------------>

e[Klocalhost*CLI>
e[0KScheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:196.36.166.174:5060 —>
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 196.36.166.174;branch=z9hG4bKb0d5.6fea1e94.0

From: “+27814488644” sip:[email protected];tag=as4c387c47

Call-ID: [email protected]

To: sip:[email protected];tag=as07970e3c

CSeq: 102 ACK

Max-Forwards: 70

User-Agent: Eyebeam

Content-Length: 0

<------------->

e[Klocalhost*CLI>
e[0K— (9 headers 0 lines) —

e[Klocalhost*CLI> exit

Executing last minute cleanups
e[0me]0;root@localhost:~a[root@localhost ~]#
e[K[root@localhost ~]# asterisk -rv
Asterisk 1.6.2.11, Copyright © 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

One of your authentication parameters in peer 3480234650 does not match so you are getting back an unauthorized message from the peer you are trying to talk to.

Hi

Thanks for this information. I have tried many things but couldnt come right. I have since created many NAT rules etc. I bypass the asterisk and configure the xlite phone it works fine.

What do you suggest?

Regards

As SkyKingOH mentioned, you have an authorization issue when it is trying to pass the call to you.

<— Reliably Transmitting (no NAT) to 196.36.166.174:5060 —>
SIP/2.0 401 Unauthorized

If it works in XLite, that means there is a configuration issue with the Trunk in Asterisk. Is there support from your SIP Provider on getting the Trunk configured?

Hi

I have sent an email to the SP and i await there reply. Thanks

Do you realize that you have not posted your trunk config?

No I havent :

Here it is:

Outbound:
Dail Rule : XXXXXXXXXX

Peer Details:
host=196.36.166.174
username=3480234650
secret=abcdefg
type=peer

Inbound:

secret=abcdefg
type=user
context=from-internal

Reg String:3480234650:[email protected]

Regards

That s not a complete comfig. I can see why it is not working.

Who is your provider? Do they provide sample Asterisk configs? (Don’t even mention FreePBX it will confuse).

Hi

I use an SP called Wanatel they are based in South Africa(http://www.wanatel.co.za) I have logged a call with them I should get some feedback on Monday. They have posted configs on there website but not for asterisk.

Regards

Hi Guys

The SP says that my server must allow G729 codec. Im not sure how to check if my server has G729 and if it doesnt have it how can i install it?

Regards

You have to purchase g.729 licenses from Digium and install them.

HI Guys

I bought a g729 codec license from Digium , How can i install this codec, I have just emailed the key

Regards

HI I seam to have the codec installed but still no incoming calls:
<— SIP read from UDP:196.36.166.174:5060 —>
OPTIONS sip:41.132.107.212:5060 SIP/2.0
Via: SIP/2.0/UDP 196.36.166.174:5060;branch=0
From: sip:[email protected];tag=4a1c3157
To: sip:41.132.107.212:5060
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Looking for s in from-sip-external (domain 41.132.107.212)

<— Transmitting (no NAT) to 196.36.166.174:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 196.36.166.174:5060;branch=0;received=196.36.166.174
From: sip:[email protected];tag=4a1c3157
To: sip:41.132.107.212:5060;tag=as2fc5465a
Call-ID: [email protected]
CSeq: 1 OPTIONS
Server: FPBX-2.7.0(1.6.2.11)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:10.0.0.6
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Reliably Transmitting (NAT) to 10.0.0.5:37584:
OPTIONS sip:[email protected]:37584;rinstance=7072a52702cd9b83 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK34a7b9d0;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as19aba9e1
To: sip:[email protected]:37584;rinstance=7072a52702cd9b83
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: FPBX-2.7.0(1.6.2.11)
Date: Mon, 17 Oct 2011 18:39:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.0.0.5:37584 —>

<------------->

<— SIP read from UDP:10.0.0.5:37584 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK34a7b9d0;rport=5060
Contact: sip:10.0.0.5:37584
To: sip:[email protected]:37584;rinstance=7072a52702cd9b83;tag=824da5c5
From: "Unknown"sip:[email protected];tag=as19aba9e1
Call-ID: [email protected]
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Really destroying SIP dialog ‘[email protected]’ Method: ACK
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 196.36.166.174:5060:
REGISTER sip:196.36.166.174 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK62de13cf;rport
Max-Forwards: 70
From: sip:[email protected];tag=as3f8186c1
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 106 REGISTER
User-Agent: FPBX-2.7.0(1.6.2.11)
Authorization: Digest username=“3480234650”, realm=“196.36.166.174”, algorithm=MD5, uri=“sip:196.36.166.174”, nonce=“4e9c76d000013af34adab1a8e24b551bd600591fc3db5d78”, response="95c9600d4f0af0c591071c7ca93cd196"
Expires: 120
Contact: sip:[email protected]
Content-Length: 0


<— SIP read from UDP:196.36.166.174:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK62de13cf;rport=5060;received=41.132.107.212
From: sip:[email protected];tag=as3f8186c1
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 106 REGISTER
Server: nice Proxy
Content-Length: 0

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

<— SIP read from UDP:196.36.166.174:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK62de13cf;rport=5060;received=41.132.107.212
From: sip:[email protected];tag=as3f8186c1
To: sip:[email protected];tag=a8e7909f4771ff08eb3d23747aa3813f.aad4
Call-ID: [email protected]
CSeq: 106 REGISTER
WWW-Authenticate: Digest realm=“196.36.166.174”, nonce=“4e9c773a00013bebfd431c8d8b35be3783e8f8b4044c5a64”, stale=true
Server: nice Proxy
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name 196.36.166.174
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 196.36.166.174:5060:
REGISTER sip:196.36.166.174 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK4a480ede;rport
Max-Forwards: 70
From: sip:[email protected];tag=as5c35f371
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 107 REGISTER
User-Agent: FPBX-2.7.0(1.6.2.11)
Authorization: Digest username=“3480234650”, realm=“196.36.166.174”, algorithm=MD5, uri=“sip:196.36.166.174”, nonce=“4e9c773a00013bebfd431c8d8b35be3783e8f8b4044c5a64”, response="68a5e19b1d32f53bd87fa6671deb4a94"
Expires: 120
Contact: sip:[email protected]
Content-Length: 0


<— SIP read from UDP:196.36.166.174:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK4a480ede;rport=5060;received=41.132.107.212
From: sip:[email protected];tag=as5c35f371
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 107 REGISTER
Server: nice Proxy
Content-Length: 0

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

<— SIP read from UDP:196.36.166.174:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.6:5060;branch=z9hG4bK4a480ede;rport=5060;received=41.132.107.212
From: sip:[email protected];tag=as5c35f371
To: sip:[email protected];tag=a8e7909f4771ff08eb3d23747aa3813f.103e
Call-ID: [email protected]
CSeq: 107 REGISTER
Contact: sip:[email protected];expires=120;received="sip:41.132.107.212:5060"
Server: nice Proxy
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:196.36.166.174:5060 —>
OPTIONS sip:41.132.107.212:5060 SIP/2.0
Via: SIP/2.0/UDP 196.36.166.174:5060;branch=0
From: sip:[email protected];tag=822c3157
To: sip:41.132.107.212:5060
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Looking for s in from-sip-external (domain 41.132.107.212)

<— Transmitting (no NAT) to 196.36.166.174:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 196.36.166.174:5060;branch=0;received=196.36.166.174
From: sip:[email protected];tag=822c3157
To: sip:41.132.107.212:5060;tag=as115de65c
Call-ID: [email protected]
CSeq: 1 OPTIONS
Server: FPBX-2.7.0(1.6.2.11)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:10.0.0.6
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
localhost*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@localhost ~]#