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rsenykoff@harrislogic.com

2004-Dec-29 07:24 UTC

[Asterisk-Users] Polycomm IP500 dropping incoming calls

Hello everyone.

I can place outgoing calls no problem with my IP500 (using teliax as our 
provider). Thing is, when a call comes in, 90% of the time when I pick up 
the handset it drops the call immediately. I turned on SIP debug, and have 
listed my extension config from sip.conf. Any help is greatly 
appreciated.... sooo close.... TIA! -Ron

[3004]
type=friend
username=3004
password=XXX
host=dynamic
;host=192.168.4.204
;host=static
dtmfmode=inband
defaultip=192.168.4.204
context=default
disallow=all
allow=ulaw
;nat=yes
callerid="George W. Bush" <3004>
mailbox=3004


SIP Debugging Enabled
    -- Accepting AUTHENTICATED call from 204.188.109.139, requested format 
= 4, actual format = 4
    -- Executing DigitTimeout("IAX2[teliax@teliax]/3", "5")
in new stack
    -- Set Digit Timeout to 5
    -- Executing ResponseTimeout("IAX2[teliax@teliax]/3",
"10") in new
stack
    -- Set Response Timeout to 10
    -- Executing Macro("IAX2[teliax@teliax]/3",
"stdexten|3004|SIP/3004")
in new stack
    -- Executing Dial("IAX2[teliax@teliax]/3",
"SIP/3004|20") in new stack
We're at 192.168.4.5 port 15760
Answering with preferred capability 4
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:3004@192.168.4.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf
From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399
To: <sip:3004@192.168.4.204>
Contact: <sip:3124048745@192.168.4.5>
Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 29 Dec 2004 20:20:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 156

v=0
o=root 1879 1879 IN IP4 192.168.4.5
s=session
c=IN IP4 192.168.4.5
t=0 0
m=audio 15760 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
 (no NAT) to 192.168.4.204:5060
    -- Called 3004


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf
From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399
To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8
CSeq: 102 INVITE
Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5
Contact: <sip:3004@192.168.4.204:5060>
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Content-Length: 0


9 headers, 0 lines


Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf
From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399
To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8
CSeq: 102 INVITE
Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5
Contact: <sip:3004@192.168.4.204:5060>
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Allow-Events: talk,hold,conference
Content-Length: 0


10 headers, 0 lines
    -- SIP/3004-5a28 is ringing


Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf
From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399
To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8
CSeq: 102 INVITE
Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5
Contact: <sip:3004@192.168.4.204:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, 
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Content-Type: application/sdp
Content-Length: 148

v=0
o=- 915180542 915180542 IN IP4 192.168.4.204
s=Polycom IP Phone
c=IN IP4 192.168.4.204
t=0 0
m=audio 2236 RTP/AVP 0
a=rtpmap:0 PCMU/8000

11 headers, 7 lines
Found audio format UNKN
Found description format PCMU
Capabilities: us - 4, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:3004@192.168.4.204:5060>
set_destination: Parsing <sip:3004@192.168.4.204:5060> for address/port to
send to
set_destination: set destination to 192.168.4.204, port 5060
Transmitting:
ACK sip:3004@192.168.4.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf
From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399
To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8
Contact: <sip:3124048745@192.168.4.5>
Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.4.204:5060
    -- SIP/3004-5a28 answered IAX2[teliax@teliax]/3
set_destination: Parsing <sip:3004@192.168.4.204:5060> for address/port to
send to
set_destination: set destination to 192.168.4.204, port 5060
Reliably Transmitting:
BYE sip:3004@192.168.4.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf
From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399
To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8
Contact: <sip:3124048745@192.168.4.5>
Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.4.204:5060
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 
'IAX2[teliax@teliax]/3' in macro 'stdexten'
  == Spawn extension (default, 9722150488, 3) exited non-zero on 
'IAX2[teliax@teliax]/3'
    -- Hungup 'IAX2[teliax@teliax]/3'


Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf
From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399
To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8
CSeq: 103 BYE
Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5
Contact: <sip:3004@192.168.4.204:5060>
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Content-Length: 0
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rsenykoff@harrislogic.com

2004-Dec-30 12:30 UTC

head link

[Asterisk-Users] Polycomm IP500 dropping incoming calls

</snip>
Hello everyone. 

I can place outgoing calls no problem with my IP500 (using teliax as our 
provider). Thing is, when a call comes in, 90% of the time when I pick up 
the handset it drops the call immediately. I turned on SIP debug, and have 
listed my extension config from sip.conf. Any help is greatly 
appreciated.... sooo close.... TIA! -Ron 
</snip>

Figured out it was a NAT issue. We were using 1-1 NAT behind a Sonicwall. 
Changed it to simple port forwarding and all is fine.

-Ron
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