I can’t make outbound calls with my Polycom 501, called extension 101, but it will accept incoming. On outgoing I get a fast busy. Asterisk works fine on another type of extension a softphone, X-Lite. The output from sip debug when I press any key on the Polycom and dial:
[code]<-- SIP read from 192.168.0.179:5060:
SUBSCRIBE sip:101@192.168.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc49dd8af7703ACD0
From: “Allan” sip:101@192.168.0.2;tag=DBD79351-CB2EA21C
To: sip:101@192.168.0.2:5060
CSeq: 1 SUBSCRIBE
Call-ID: e5a10655-e8eca00b-a008124e@192.168.0.179
Contact: sip:101@192.168.0.179
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: line-seize
Call-Info: sip:192.168.0.2;appearance-index=1
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Max-Forwards: 70
Expires: 30
Content-Length: 0
— (14 headers 0 lines)—
Creating new subscription
Sending to 192.168.0.179 : 5060 (no NAT)
Found peer '101’
Transmitting (no NAT) to 192.168.0.179:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc49dd8af7703ACD0;received=192.168.0.179
From: “Allan” sip:101@192.168.0.2;tag=DBD79351-CB2EA21C
To: sip:101@192.168.0.2:5060;tag=as0462fab2
Call-ID: e5a10655-e8eca00b-a008124e@192.168.0.179
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:101@192.168.0.2
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="76bda117"
Content-Length: 0
Scheduling destruction of SIP dialog ‘e5a10655-e8eca00b-a008124e@192.168.0.179’ in 32000 ms (Method: SUBSCRIBE)
australia*CLI>
<-- SIP read from 192.168.0.179:5060:
SUBSCRIBE sip:101@192.168.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bK54bdcfbaA07B31A7
From: “Allan” sip:101@192.168.0.2;tag=DBD79351-CB2EA21C
To: sip:101@192.168.0.2:5060
CSeq: 2 SUBSCRIBE
Call-ID: e5a10655-e8eca00b-a008124e@192.168.0.179
Contact: sip:101@192.168.0.179
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: line-seize
Call-Info: sip:192.168.0.2;appearance-index=1
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Authorization: Digest username=“101”, realm=“asterisk”, nonce=“76bda117”, uri=“sip:101@192.168.0.2:5060”, response=“6066c83e2faef5eb1c782d3d59213292”, algorithm=MD5
Max-Forwards: 70
Expires: 30
Content-Length: 0
— (15 headers 0 lines)—
Found peer '101’
Looking for 101 in from-101 (domain 192.168.0.2)
Transmitting (no NAT) to 192.168.0.179:5060:
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bK54bdcfbaA07B31A7;received=192.168.0.179
From: “Allan” sip:101@192.168.0.2;tag=DBD79351-CB2EA21C
To: sip:101@192.168.0.2:5060;tag=as0462fab2
Call-ID: e5a10655-e8eca00b-a008124e@192.168.0.179
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:101@192.168.0.2
Content-Length: 0[/code]
And the pertinent information from extensions.conf
[from-101]
exten => _X.,1,Answer
exten => _X.,2,Wait(2)
exten => _X.,3,Playback(tt-monkeys)
exten => _X.,4,Hangup
exten => 200,1,Dial(SIP/200,,r)
and sip.conf
[101]
username=101
type=friend
secret=4234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=never
mailbox=101@device
host=dynamic
dtmfmode=rfc2833
context=from-101
canreinvite=no
callerid=Allan Wilson <101>
It shows the phone is communicating correctly
australia*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
200/200 192.168.0.158 D 8922 OK (102 ms)
101/101 192.168.0.179 D 5060 OK (70 ms)
2 sip peers [2 online , 0 offline]
In the sip settings on the Polycom I have the ip address of asterisk as the host and 5060 for the port and have the same things set for outbound settings. Any ideas or examples of others using this phone. Thanks for any help.
Allan