[Asterisk-Users] Messengers calls dropped (SIP problem?)

Sat Apr 24 05:27:39 MST 2004

I have asterisk with following users;

a) zaphfc ISDN card with two channels
b) two mediatrix FXS gateways with four channels each
c) 1x CISCO 7905G
d) two notebooks with MS Messenger 4.7

Now, it seems that any combination works correctly in all combinations
except when I call from MS messenger and then call is dropped always in
25th second of the call. Any ideas what I did wrong?

here is my messenger sip.conf portion;

[marko]
type=friend
reinvite=no
username=marko
host=dynamic
mailbox=1300

here is my cisco 7905 sip.conf portion;

[123]
type=peer
reinvite=no
callerid= "Marko Rakar"
username=123
secret=1234
dtmfmode=inband
careinvite=yes
host=dynamic
defaultip=192.168.3.52
incominglimit=2
outgoinglimit=2

here is a part of my sip debug file

9 headers, 0 lines
Sending to 192.168.3.54 : 14250 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.54:14250
From: "marko"
<sip:marko at asterisk>;tag=124ece30-95ed-4a45-8a62-5bacd517e1ae
To: <sip:1361 at asterisk;user=phone>;tag=as05865310
Call-ID: 860136be-1ae5-44db-b86d-90b5d31f0c08 at 192.168.3.54
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1361 at 192.168.3.6>
Content-Length: 0

 to 192.168.3.54:14250
set_destination: Parsing
<sip:123 at 192.168.3.52:5060;user=phone;transport=udp> for address/port to
send to
set_destination: set destination to 192.168.3.52, port 5060
We're at 192.168.3.6 port 29312
Answering with preferred capability 4
Answering with non-codec capability 1
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:123 at 192.168.3.52:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235
From: "marko" <sip:asterisk at 192.168.3.6>;tag=as33a10ddc
To: <sip:123 at 192.168.3.52>;tag=1930002232
Contact: <sip:asterisk at 192.168.3.6>
Call-ID: 097749b00d1b935e68869c241d350bd8 at 192.168.3.6
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 5963 5965 IN IP4 192.168.3.6
s=session
c=IN IP4 192.168.3.6
t=0 0
m=audio 29312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 192.168.3.52:5060

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235
From: "marko" <sip:asterisk at 192.168.3.6>;tag=as33a10ddc
To: <sip:123 at 192.168.3.52>;tag=1930002232
Call-ID: 097749b00d1b935e68869c241d350bd8 at 192.168.3.6
CSeq: 104 INVITE
Contact: 123 <sip:123 at 192.168.3.52:5060;user=phone;transport=udp>
Server: Cisco-CP7905/1.01-030807A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 202
Content-Type: application/sdp

v=0
o=123 37649 37649 IN IP4 192.168.3.52
s=Cisco 7905 SIP Call
c=IN IP4 192.168.3.52
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

11 headers, 9 lines
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format telephone-event
Capabilities: us - 12, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: <sip:123 at 192.168.3.52:5060;user=phone;transport=udp>
set_destination: Parsing
<sip:123 at 192.168.3.52:5060;user=phone;transport=udp> for address/port to
send to
set_destination: set destination to 192.168.3.52, port 5060
Transmitting:
ACK sip:123 at 192.168.3.52:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235
From: "marko" <sip:asterisk at 192.168.3.6>;tag=as33a10ddc
To: <sip:123 at 192.168.3.52>;tag=1930002232
Contact: <sip:asterisk at 192.168.3.6>
Call-ID: 097749b00d1b935e68869c241d350bd8 at 192.168.3.6
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.3.52:5060
set_destination: Parsing
<sip:123 at 192.168.3.52:5060;user=phone;transport=udp> for address/port to
send to
set_destination: set destination to 192.168.3.52, port 5060
Reliably Transmitting:
BYE sip:123 at 192.168.3.52:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235
From: "marko" <sip:asterisk at 192.168.3.6>;tag=as33a10ddc
To: <sip:123 at 192.168.3.52>;tag=1930002232
Contact: <sip:asterisk at 192.168.3.6>
Call-ID: 097749b00d1b935e68869c241d350bd8 at 192.168.3.6
CSeq: 105 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.3.52:5060

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235
From: "marko" <sip:asterisk at 192.168.3.6>;tag=as33a10ddc
To: <sip:123 at 192.168.3.52>;tag=1930002232
Call-ID: 097749b00d1b935e68869c241d350bd8 at 192.168.3.6
CSeq: 105 BYE
Server: Cisco-CP7905/1.01-030807A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0

9 headers, 0 lines

----
Give someone a fish, you feed him for one day. Teach him how to fish,
and you lose a steady customer.

mailto:marko at printel.hr
http://printel.hr  