Greetings to All!
Help to solve a problem:
There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate
800 2FXS).
In sip.conf they are registered so:
[3301]
type=friend
host=172.222.135.11
username=3301
secret=0000
defaultip=172.222.135.11
dtmfmode=rfc2833
context=it
callerid="VoIPGate2Line1" <3301>
allow=g723.1
[3302]
type=friend
host=172.222.135.11
username=3302
secret=0000
defaultip=172.222.135.11
dtmfmode=rfc2833
context=it
callerid="VoIPGate2Line2" <3302>
allow=g723.1
[3440]
type=friend
host=dynamic
username=3440
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine1" <3440>
allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
[3441]
type=friend
host=dynamic
username=3441
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine2" <3441>
allow=g723.1
[3442]
type=friend
host=dynamic
username=3442
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine3" <3442>
allow=g723.1
[3443]
type=friend
host=dynamic
username=3443
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine4" <3443>
allow=g723.1
When I call from internal telephone to some of this numbers - the call
goes, but when I pickup phone - communication breaks...
Please, help to understand!
Here a log:
<-- SIP read from 172.222.135.11:5060:
SIP/2.0 200 OK
From: "Unknown"<sip:[EMAIL PROTECTED]>;tag=as2335e618
To: <sip:[EMAIL PROTECTED]>;tag=ac16230b-13c4-ad-2c457-4966
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.11.0.150:5060;rport=5060;branch=z9hG4bK0e38ec58
Supported: replaces
User-Agent: FXS_GW (2sipfxs.112)
Contact: <sip:[EMAIL PROTECTED]:0>
Content-Type: application/sdp
Content-Length: 220
v=0
o=FXS_GW 12367 0 IN IP4 172.222.135.11
s=Audio Session
i=Audio Session
c=IN IP4 172.222.135.11
t=0 0
m=audio 0 RTP/AVP 4 101
a=ptime:30
a=fmtp:101 0-11
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
--- (11 headers 11 lines)---
Found RTP audio format 4
Found RTP audio format 101
Peer video RTP is at port 172.222.135.11:65535
Found description format G723
Found description format telephone-event
Capabilities: us - 0x8010f (g723|gsm|ulaw|alaw|g729|h263), peer -
audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:[EMAIL PROTECTED]:0> for address/port to
send to
set_destination: set destination to 172.222.135.11, port 0
Transmitting (no NAT) to 172.222.135.11:0:
ACK sip:[EMAIL PROTECTED]:0 SIP/2.0
Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK3f42da36;rport
From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as2335e618
To: <sip:[EMAIL PROTECTED]>;tag=ac16230b-13c4-ad-2c457-4966
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Oct 30 08:15:34 WARNING[19154]: chan_sip.c:1082 __sip_xmit: sip_xmit of
0x6f39d3c8 (len 387) to 172.222.135.11:0 returned -1: Invalid argument
Retransmitting #4 (no NAT) to 172.222.135.11:0:
BYE sip:[EMAIL PROTECTED]:0 SIP/2.0
Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK4ee204c1;rport
From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as2335e618
To: <sip:[EMAIL PROTECTED]>;tag=ac16230b-13c4-ad-2c457-4966
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Oct 30 08:15:35 WARNING[19154]: chan_sip.c:1082 __sip_xmit: sip_xmit of
0x8151630 (len 387) to 172.222.135.11:0 returned -1: Invalid argument
Destroying call '[EMAIL PROTECTED]'
<-- SIP read from 172.222.135.11:5060:
BYE sip:[EMAIL PROTECTED] SIP/2.0
From: <sip:[EMAIL PROTECTED]>;tag=ac16230b-13c4-ad-2c457-4966
To: "Unknown"<sip:[EMAIL PROTECTED]>;tag=as2335e618
Call-ID: [EMAIL PROTECTED]
CSeq: 1 BYE
Via: SIP/2.0/UDP 172.222.135.11:5060;branch=z9hG4bK-c0-30e94-d54
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (2sipfxs.112)
Content-Length: 0
--- (10 headers 0 lines)---
Sending to 172.222.135.11 : 5060 (non-NAT)
Transmitting (no NAT) to 172.222.135.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.222.135.11:5060;branch=z9hG4bK-c0-30e94-d54;received=172.222.135.11
From: <sip:[EMAIL PROTECTED]>;tag=ac16230b-13c4-ad-2c457-4966
To: "Unknown"<sip:[EMAIL PROTECTED]>;tag=as2335e618
Call-ID: [EMAIL PROTECTED]
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
Retransmitting #5 (no NAT) to 172.222.135.11:5060:
BYE sip:[EMAIL PROTECTED]:0 SIP/2.0
Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK4ee204c1;rport
From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as2335e618
To: <sip:[EMAIL PROTECTED]>;tag=ac16230b-13c4-ad-2c457-4966
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<-- SIP read from 172.222.135.11:5060:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
From: "Unknown"<sip:[EMAIL PROTECTED]>;tag=as2335e618
To: <sip:[EMAIL PROTECTED]>;tag=ac16230b-13c4-ad-2c457-4966
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.11.0.150:5060;rport=5060;branch=z9hG4bK4ee204c1
Supported: replaces
Content-Length: 0
begin:vcard
fn:Eugeniy Khvastunov
n:Khvastunov;Eugeniy
org:Digma;IT
adr:;;;Kharkov;Kh;;Ukraine
email;internet:[EMAIL PROTECTED]
title:System Administrator
tel;work:+380675745646
tel;cell:+380504063116
version:2.1
end:vcard
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