[Asterisk-Users] Polycom V500 With Asterisk Setup

[David Choo]

Dear All,

We've got 1 set of Polycom V500 Video Conferencing Kit in my Office. I'm
trying to link it with Asterisk and is facing some issues. Would like to
seek your kind advise.

The Polycom V500 is unable to make the outgoing calls, and will always
report the "ENTER ERROR HERE".

"sip show peers" does not shows that the Polycom V500 being able to
register. The account is working alright as I've used the account on
Eyebeam and its working fine.

Here are the debug logs for the System

<-- SIP read from 192.168.100.146:5060:
INVITE sip:404 at 192.168.100.146 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655
Max-Forwards: 70
From: <sip:898 at 192.168.100.146>;epid=82042503E72EB0;tag=df8c4526
To: <sip:404 at 192.168.100.146>
Call-ID: e8c14000 at 192.168.100.146
CSeq: 1 INVITE
User-Agent: Polycom V500 Release 7.5 - 15Dec2004 10:12
Contact: <sip:192.168.100.146>
Content-Type: application/sdp
Content-Length: 899

v=0
o=Vigor11 1627471320 0 IN IP4 192.168.100.146
s=-
c=IN IP4 192.168.100.146
b=AS:384
t=0 0
m=audio 49178 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18
a=rtpmap:99 SIREN14/16000
a=fmtp:99 bitrate=48000
a=rtpmap:98 SIREN14/16000
a=fmtp:98 bitrate=32000
a=rtpmap:97 SIREN14/16000
a=fmtp:97 bitrate=24000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=24000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=16000
a=rtpmap:9 G722/8000
a=rtpmap:15 G728/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729A/8000
m=video 49180 RTP/AVP 109 34 96 31
b=TIAS:384000
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800c max-mbps=10000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 F J T
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1 QCIF=1
m=data 49182 RTP/AVP 100
a=rtpmap:100 H224

--- (11 headers 35 lines)---
Using latest request as basis request
Sending to 192.168.100.146 : 5060 (non-NAT)
Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5947 check_user_full: Setting NAT
on RTP to 524288
Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5951 check_user_full: Setting NAT
on VRTP to 524288
Reliably Transmitting (NAT) to 192.168.100.146:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.146;branch=z9hG4bK1f784655;received=192.168.100.146;rport=5060
From: <sip:898 at 192.168.100.146>;epid=82042503E72EB0;tag=df8c4526
To: <sip:404 at 192.168.100.146>;tag=as36644353
Call-ID: e8c14000 at 192.168.100.146
CSeq: 1 INVITE
User-Agent: nVoice PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:404 at 61.14.78.47>
Proxy-Authenticate: Digest realm="nvoice", nonce="60b31ab3"
Content-Length: 0


---
Scheduling destruction of call 'e8c14000 at 192.168.100.146' in 15000 ms
Found user '898'
tannery*CLI>
<-- SIP read from 192.168.100.146:5060:
ACK sip:192.168.100.146 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655
Max-Forwards: 70
From: <sip:898 at 192.168.100.146>;epid=82042503E72EB0;tag=df8c4526
To: <sip:404 at 192.168.100.146>
Call-ID: e8c14000 at 192.168.100.146
CSeq: 1 ACK
Contact: <sip:192.168.100.146>
Content-Length: 0



Best Regards,

==============================
David Choo
Systems Engineer
Business & Technology Division
"Engineered for Changing Businesses"
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :DavidChoo at Espore.com
=============================

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