Incomming call from extention , not from asterisk!

Asterisk Asterisk Support
1

Hi ,
i have remote asterisk server

if i use my zoiper soft phone and put usr/pwd and ip

i can call out and recivie calls on my extention

now im trying to use this extention on my asterisk o be able to send and recive calls using asterisk

so what i have … i can only call out but i cant recive any calls in asterisk

i enabled debug and sip debug

there is no thing about the incomming call

only debug apperaes when we do out calls …

any help about that ??

here is my info

sip.conf

[icosnet]
host=196.41.228.32
defaultuser=213932400845
secret=1aby4elztj7lw845
type=peer
context=from-internal

[root@localhost ~]# cat /etc/asterisk/sip_registrations.conf
register=213982480845:1aby4eiztj7lw845@196.41.228.32

[root@localhost ~]# asterisk -rvv
Asterisk 11.16.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
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.

Connected to Asterisk 11.16.0 currently running on localhost (pid = 1993)
localhostCLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
icosnet/213982400845 196.41.228.32 Yes Yes 5060 Unmonitored
16 sip peers [Monitored: 4 online, 5 offline Unmonitored: 5 online, 2 offline]
localhostCLI>

any help ???

2

There is too little information to debug this, although publishing the credentials for a device (not extension) that has from_internal level access is not a good idea.

Turn up verbosity to at least 5, so that you can see the dialplan executing for the incoming call, and use sip set debug on.

When getting peer status, provide the full status, not just the summary lines.

3

OK David , done

i setup sip debug and core verbose to 5

here is the logs :

<— SIP read from UDP:196.41.228.32:5060 —>
INVITE sip:s@10.23.10.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 196.41.228.32:5060;branch=z9hG4bK-524287-1—ee07723238625f65;rport
Via: SIP/2.0/UDP 196.41.228.32:5061;branch=z9hG4bK-rrgi5b3dlkwubclb;rport=5061
Max-Forwards: 69
Record-Route: sip:196.41.228.32:5060;lr;transport=UDP
Contact: "Anonymous"sip:196.41.228.32:5061
To: sip:213982400845@196.41.228.32
From: sip:0550555069@196.41.228.32;tag=a7jvvlq4r55wqtfx.o
Call-ID: e9ce15faefb36f29@196.41.228.244~2o
CSeq: 383 INVITE
Expires: 300
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
P-Asserted-Identity: sip:0550555069@196.41.228.32
Remote-Party-ID: sip:0550555069@196.41.228.32;party=calling
h323-conf-id: 573029038-707663210-3735624150-3088455831
Portasip-3264-action: offer 1
cisco-GUID: 573029038-707663210-3735624150-3088455831
Content-Length: 305

v=0
o=Sippy 4447977293973225227 1 IN IP4 196.41.228.32
s=-
t=0 0
m=audio 60884 RTP/AVP 8 0 18 3 4 101
c=IN IP4 196.41.228.32
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
<------------->
— (21 headers 13 lines) —
Sending to 196.41.228.32:5060 (NAT)
Sending to 196.41.228.32:5060 (NAT)
Using INVITE request as basis request - e9ce15faefb36f29@196.41.228.244~2o
Found peer ‘from-internal’ for ‘0550555069’ from 196.41.228.32:5060

<— Reliably Transmitting (NAT) to 196.41.228.32:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 196.41.228.32:5060;branch=z9hG4bK-524287-1—ee07723238625f65;received=196.41.228.32;rport=5060
Via: SIP/2.0/UDP 196.41.228.32:5061;branch=z9hG4bK-rrgi5b3dlkwubclb;rport=5061
From: sip:0550555069@196.41.228.32;tag=a7jvvlq4r55wqtfx.o
To: sip:213982400845@196.41.228.32;tag=as4f38d300
Call-ID: e9ce15faefb36f29@196.41.228.244~2o
CSeq: 383 INVITE
Server: FPBX-AsteriskNOW-12.0.70(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="42b62c8a"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘e9ce15faefb36f29@196.41.228.244~2o’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:196.41.228.32:5060 —>
ACK sip:s@10.23.10.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 196.41.228.32:5060;branch=z9hG4bK-524287-1—ee07723238625f65;rport
Max-Forwards: 70
To: sip:213982400845@196.41.228.32;tag=as4f38d300
From: sip:0550555069@196.41.228.32;tag=a7jvvlq4r55wqtfx.o
Call-ID: e9ce15faefb36f29@196.41.228.244~2o
CSeq: 383 ACK
Content-Length: 0

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

<— SIP read from UDP:196.41.228.32:5060 —>

4

The IP address in the INVITE header is wrong. As the source is on a public IP address, one would expect a public address as the destination. However, Asterisk will normally ignore this.

The trace is either incomplete, or the device at 196.41.228.32 has not been configured with any authorisation data.

5

hi david this is the full trace

agian

which ip is wrong in the header and how to correct it ?

is it 196.41.228.32 (my peer trunk ) ??
or
196.41.228.244
???

agian , if i replace the asterisk with zoiper softphone with the same network settings , im able to send and receive calls !!!

any explanation for that ?

thanks agian Mr david

cheers

6

[quote]SIP/2.0 401 Unauthorized
[/quote]
I think he need to add the remotesecret parameter or set the old hack insecure=invite in his trunk configuration

7

As I understand this, the remote entity is a phone, not an ITSP.

8

The address that is wrong is10.23.10.2。 However this is only a technicality, as Asterisk will ignore it in this context.

The trace should have continued with a second INVITE containing authentication data based on the password. As it doesn’t, it means that the device has not been configured with a password to use.