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G.Marshall
2005-May-13 18:42 UTC
[Asterisk-Users] asterisk dials random number when receiving incoming call

Hello,

I have found asterisk is dialing a random number when it recieves a call,
would anyone know why?  The first thing I noticed found peer 4563 (this is
a n Xlite Client)


Many thanks,

Spencer

SIP Debugging Enabled
spitfire*CLI>
<-- SIP read from 82.70.154.145:5060:
INVITE sip:448715046363@iptel.tgfslp.dalmany.co.uk SIP/2.0
Max-Forwards: 10
Record-Route: <sip:82.70.154.145;ftag=as3606b893;lr=on>
Via: SIP/2.0/UDP 82.70.154.145;branch=z9hG4bK2922.6c170001.0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" <sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>
Contact: <sip:Unavailable@213.166.5.129>
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 102 INVITE
User-Agent: MSS VoIP Gateway
Date: Sat, 14 May 2005 01:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 31661 31661 IN IP4 213.166.5.129
s=session
c=IN IP4 213.166.5.129
t=0 0
m=audio 14474 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (15 headers 12 lines)---
Using latest request as basis request
Sending to 82.70.154.145 : 5060 (NAT)
Found peer '4563'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.129:14474
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 448715046363 in local-sip
list_route: hop: <sip:82.70.154.145;ftag=as3606b893;lr=on>
list_route: hop: <sip:Unavailable@213.166.5.129>
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" <sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:448715046363@82.70.154.145:5061>
Content-Length: 0


---
    -- Executing Dial("SIP/4563-5e36",
"SIP/448715046363@192.168.4.5:5061|60|r")
We're at 192.168.4.3 port 35002
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.4.5:5061:
INVITE sip:448715046363@192.168.4.5:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
From: "unknown" <sip:asterisk@192.168.4.3:5061>;tag=as60a4b224
To: <sip:448715046363@192.168.4.5:5061>
Contact: <sip:asterisk@192.168.4.3:5061>
Call-ID: 5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 14 May 2005 01:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 8318 8318 IN IP4 192.168.4.3
s=session
c=IN IP4 192.168.4.3
t=0 0
m=audio 35002 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 448715046363@192.168.4.5:5061
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 100 Trying
To: <sip:448715046363@192.168.4.5:5061>
From: "unknown" <sip:asterisk@192.168.4.3:5061>;tag=as60a4b224
Call-ID: 5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0


--- (8 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 180 Ringing
To: <sip:448715046363@192.168.4.5:5061>;tag=d416591c6d2e2378i1
From: "unknown" <sip:asterisk@192.168.4.3:5061>;tag=as60a4b224
Call-ID: 5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0


--- (8 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 200 OK
To: <sip:448715046363@192.168.4.5:5061>;tag=d416591c6d2e2378i1
From: "unknown" <sip:asterisk@192.168.4.3:5061>;tag=as60a4b224
Call-ID: 5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Contact: PSTN Line <sip:448715046363@192.168.4.5:5061>
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 233
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
ontent-Type: application/sdp

v=0
o=- 3069797 3069797 IN IP4 192.168.4.5
s=-
c=IN IP4 192.168.4.5
t=0 0
m=audio 16452 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.4.5:16452
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:448715046363@192.168.4.5:5061>
set_destination: Parsing <sip:448715046363@192.168.4.5:5061> for
address/port to send to
set_destination: set destination to 192.168.4.5, port 5061
Transmitting (no NAT) to 192.168.4.5:5061:
ACK sip:448715046363@192.168.4.5:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK660ef268
From: "unknown" <sip:asterisk@192.168.4.3:5061>;tag=as60a4b224
To: <sip:448715046363@192.168.4.5:5061>;tag=d416591c6d2e2378i1
Contact: <sip:asterisk@192.168.4.3:5061>
Call-ID: 5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" <sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>;tag=as7681341a
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:448715046363@82.70.154.145:5061>
Content-Length: 0


---
    -- SIP/192.168.4.5:5061-05b4 is ringing
    -- SIP/192.168.4.5:5061-05b4 answered SIP/4563-5e36
We're at 82.70.154.145 port 35040
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with capability 0x100 (g729)
Answering with capability 0x400 (ilbc)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
Record-Route: <sip:82.70.154.145;ftag=as3606b893;lr=on>
From: "unknown" <sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>;tag=as7681341a
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:448715046363@82.70.154.145:5061>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 8318 8318 IN IP4 82.70.154.145
s=session
c=IN IP4 82.70.154.145
t=0 0
m=audio 35040 RTP/AVP 3 0 8 18 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/4563-5e36 and
SIP/192.168.4.5:5061-05b4
spitfire*CLI>
<-- SIP read from 82.70.154.145:5060:
ACK sip:448715046363@82.70.154.145:5061 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 82.70.154.145;branch=0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK538adb90
From: "unknown" <sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>;tag=as7681341a
Contact: <sip:Unavailable@213.166.5.129>
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 102 ACK
User-Agent: MSS VoIP Gateway
Content-Length: 0
P-hint: rr-enforced


--- (12 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 82.70.154.145:5060:
BYE sip:448715046363@82.70.154.145:5061 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 82.70.154.145;branch=z9hG4bK3922.c51fd76.0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK4eaf2728
From: "unknown" <sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>;tag=as7681341a
Contact: <sip:Unavailable@213.166.5.129>
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 103 BYE
User-Agent: MSS VoIP Gateway
Content-Length: 0
Route: <sip:448715046363@82.70.154.145:5061>
P-hint: rr-enforced


--- (13 headers 0 lines)---
Sending to 82.70.154.145 : 5060 (NAT)
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK3922.c51fd76.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK4eaf2728
From: "unknown" <sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>;tag=as7681341a
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:448715046363@82.70.154.145:5061>
Content-Length: 0


---
set_destination: Parsing <sip:448715046363@192.168.4.5:5061> for
address/port to send to
set_destination: set destination to 192.168.4.5, port 5061
Reliably Transmitting (no NAT) to 192.168.4.5:5061:
BYE sip:448715046363@192.168.4.5:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK2f906ce7
From: "unknown" <sip:asterisk@192.168.4.3:5061>;tag=as60a4b224
To: <sip:448715046363@192.168.4.5:5061>;tag=d416591c6d2e2378i1
Contact: <sip:asterisk@192.168.4.3:5061>
Call-ID: 5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0


---
  == Spawn extension (local-sip, 448715046363, 1) exited non-zero on
'SIP/4563-5e36'
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 200 OK
To: <sip:448715046363@192.168.4.5:5061>;tag=d416591c6d2e2378i1
From: "unknown" <sip:asterisk@192.168.4.3:5061>;tag=as60a4b224
Call-ID: 5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK2f906ce7
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0


--- (8 headers 0 lines)---
Destroying call '5745d9355c8f8fb22349fd9f19b6b48a@192.168.4.3'
Destroying call '1f9465ed1482e9804b089a351a4174a4@213.166.5.129'
spitfire*CLI> sip no debug
SIP Debugging Disabled
spitfire*CLI>
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