Problem with Ivcomming calls

muertesbg

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#1
Hello,
I have last version of elastix.
The outgoing calls is OK, but I have a trouble with Incomming.
The asterisk and X-lite soft phone is behind NAT.
IP address of the Asterisk server is 10.0.0.240/255.255.255.0
In my sip_general_custom.conf I add these lines:
Code:
externhost = XXX.XXX.XXX.XXX ; My External IP 
externrefresh = 60
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
localnet = 10.0.0.0/255.255.255.0
Here is a snapshot of the Trunk and Inbounde Route:
Picture 1
Picture 2
When I try to make a call from outside telephone, in my Asterisk*CLI nothing is comming.An other side is busy tone.
Please help.
After enable sip debug for peer:
Code:
<--- SIP read from 212.116.145.19:5060 --->
INVITE sip:s@XXX.XXX.XXX.XXX SIP/2.0
Record-Route: <sip:212.116.145.19;lr=on;ftag=5B282A04-3E0;nat=yes>
Via: SIP/2.0/UDP 212.116.145.19;branch=z9hG4bKedf6.0c327f7.0
Via: SIP/2.0/UDP 212.116.145.6:5060;rport=52681;branch=z9hG4bKBDD881534
Remote-Party-ID: "35932XXXXXX " <sip:35932XXXXXX@ITDNET>;party=calling;screen=no;privacy=off
From: "359XXXXXXX " <sip:35932XXXXX@ITDNET>;tag=5B282A04-3E0
To: <sip:35932XXXXXX@212.116.145.19>
Date: Mon, 01 Jun 2009 13:27:30 GMT
Call-ID: CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 3400172442-1306923486-3028413264-2550145183
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1243862850
Contact: <sip:35932XXXXXXX@212.116.145.6:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 67
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 330

v=0
o=CiscoSystemsSIP-GW-UserAgent 7744 2716 IN IP4 212.116.145.6
s=SIP Call
c=IN IP4 212.116.145.6
t=0 0
m=audio 18316 RTP/AVP 8 0 18 101
c=IN IP4 212.116.145.6
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=direction:passive

<------------->
--- (23 headers 14 lines) ---
Sending to 212.116.145.19 : 5060 (no NAT)
Using INVITE request as basis request - CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
Found peer 'CoolBox511'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 212.116.145.6:18316
Found audio description format PCMA for ID 8
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=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 212.116.145.6:18316
Looking for s in from-CoolBox511 (domain XXX.XXX.XXX.XXX (My IP))

<--- Reliably Transmitting (no NAT) to 212.116.145.19:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.116.145.19;branch=z9hG4bKedf6.0c327f7.0;received=212.116.145.19
Via: SIP/2.0/UDP 212.116.145.6:5060;rport=52681;branch=z9hG4bKBDD881534
From: "35932XXXXXX " <sip:35932XXXXXX@ITDNET>;tag=5B282A04-3E0
To: <sip:35932XXXXXX@212.116.145.19>;tag=as35ad7e9c
Call-ID: CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET' in 32000 ms (Method: INVITE)
ubuntu*CLI>
<--- SIP read from 212.116.145.19:5060 --->
ACK sip:s@XXX.XXX.XXX.XXX (My IP) SIP/2.0
Via: SIP/2.0/UDP 212.116.145.19;branch=z9hG4bKedf6.0c327f7.0
From: "35932XXXXXX " <sip:35932XXXXXX@ITDNET>;tag=5B282A04-3E0
Call-ID: CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
To: <sip:35932XXXXXX@212.116.145.19>;tag=as35ad7e9c
CSeq: 101 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET' Method: ACK
In my log file is:
Code:
[Jun  1 16:26:18] NOTICE[5310] chan_sip.c: Call from '35932XXXXXX' to extension 's' rejected because extension not found.
 

muertesbg

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#2
The Problem is SOLVED.
 

ramoncio

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#3
Could you explain how you did it?
Maybe it can help some others..
 

muertesbg

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#4
Yes of course.My mistake.
In the Peer Details change context=from-trunk, also in the User Details again context=from trunk.
This is solve my problems.Now Incoming and Outgoig calls working OK.
Thank You very much for Your help.
 

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