Messag Waiting Indicator on Polycom 501

Asterisk Asterisk Support
1

I am unable to get the MWI on my polycom 501 phones to work. I have set the callback mode to contact, and the bypass instant message to enabled, but nothing seems to work. Is there something I am missing in this?

Thanks.

2

just to check the obvious you did set mailbox= in its sip.conf entry, yes?

try to capture a sip debug of * giving it a MWI notify (should happen right after it registers) and see if that has any clues…

3

Yep, the mailbox is set to [mailbox]@[context] , in that case 201@default

I didn’t see anything odd in the debug:
this is what it looked like:

<-- SIP read from 10.1.50.110:5060:
REGISTER sip:xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.50.110;branch=z9hG4bKcde75f6bBFF513B4
From: "101" <sip:101@xxx>;tag=C95760A0-3FF089C7
To: <sip:101@xxx>
CSeq: 308 REGISTER
Call-ID: 5950bc84-9bc23bba-2227f969@10.1.50.110
Contact: <sip:101@10.1.50.110>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Authorization: Digest username="101", realm="asterisk", nonce="4ebe24e9", uri="sip:xxx:5060", response="f114daf236ae4fae19d4cfc311b14aba", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.50.110 : 5060 (non-NAT)
    -- SIP Seeding peer from astdb: '101' at 101@10.1.50.110:5060 for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.50.110:5060:
OPTIONS sip:101@10.1.50.110 SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK6089f520;rport
From: "asterisk" <sip:asterisk@xxx>;tag=as40e0304f
To: <sip:101@10.1.50.110>
Contact: <sip:asterisk@xxx>
Call-ID: 318e4a1265b7736628e370955e36e43f@xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 08 Jan 2007 02:12:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

any ideas? I don’t even see anything mentioning MWI?

Thanks again

4

anything else? as i recall it should after that send a NOTIFY for mwi…

5

the entire thing for that extension that is printed out upon bootup is:

--- (10 headers 0 lines)---
Destroying call '1530bef47531c8db687ae9805415e2d4@xxx'
tel*CLI>
<-- SIP read from 10.1.50.110:5060:
REGISTER sip:xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.50.110;branch=z9hG4bK58a132c56B0D24EE
From: "201" <sip:201@xxx>;tag=5F13B47-642AE67A
To: <sip:201@xxx>
CSeq: 1 REGISTER
Call-ID: 17f4ccb-2adb0721-fbc9ee6c@10.1.50.110
Contact: <sip:201@10.1.50.110>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Max-Forwards: 70
Expires: 3600
Content-Length: 0


--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.50.110 : 5060 (non-NAT)
    -- SIP Seeding peer from astdb: '201' at 201@10.1.50.110:5060 for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.50.110:5060:
OPTIONS sip:201@10.1.50.110 SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK71cf42c0;rport
From: "asterisk" <sip:asterisk@xxx>;tag=as6c57d414
To: <sip:201@10.1.50.110>
Contact: <sip:asterisk@xxx>
Call-ID: 067237ab3aa575234a1cb3bb1dc25f8b@xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 08 Jan 2007 06:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Transmitting (no NAT) to 10.1.50.110:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.50.110;branch=z9hG4bK58a132c56B0D24EE;received=10.1.50.110
From: "201" <sip:201@xxx>;tag=5F13B47-642AE67A
To: <sip:201@xxx>
Call-ID: 17f4ccb-2adb0721-fbc9ee6c@10.1.50.110
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:201@xxx>
Content-Length: 0


---
Transmitting (no NAT) to 10.1.50.110:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.50.110;branch=z9hG4bK58a132c56B0D24EE;received=10.1.50.110
From: "201" <sip:201@xxx>;tag=5F13B47-642AE67A
To: <sip:201@xxx>;tag=as3dd7fe57
Call-ID: 17f4ccb-2adb0721-fbc9ee6c@10.1.50.110
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:201@xxx>
WWW-Authenticate: Digest realm="asterisk", nonce="12e55ba2"
Content-Length: 0


---
Scheduling destruction of call '17f4ccb-2adb0721-fbc9ee6c@10.1.50.110' in 15000 ms
Destroying call '067237ab3aa575234a1cb3bb1dc25f8b@xxx'
tel*CLI>
<-- SIP read from 10.1.50.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK71cf42c0;rport
From: "asterisk" <sip:asterisk@xxx>;tag=as6c57d414
To: <sip:201@10.1.50.110>;tag=21C3A446-BAA616BD
CSeq: 102 OPTIONS
Call-ID: 067237ab3aa575234a1cb3bb1dc25f8b@xxx
Contact: <sip:201@10.1.50.110>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Length: 0


--- (10 headers 0 lines)---
Destroying call '067237ab3aa575234a1cb3bb1dc25f8b@xxx'
tel*CLI>
<-- SIP read from 10.1.50.110:5060:
REGISTER sip:xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.50.110;branch=z9hG4bKeb17b034FEA2504
From: "201" <sip:201@xxx>;tag=5F13B47-642AE67A
To: <sip:201@xxx>
CSeq: 2 REGISTER
Call-ID: 17f4ccb-2adb0721-fbc9ee6c@10.1.50.110
Contact: <sip:201@10.1.50.110>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Authorization: Digest username="201", realm="asterisk", nonce="12e55ba2", uri="sip:xxx:5060", response="3860e56610197be73045af959b9966d3", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.50.110 : 5060 (non-NAT)
    -- SIP Seeding peer from astdb: '201' at 201@10.1.50.110:5060 for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.50.110:5060:
OPTIONS sip:201@10.1.50.110 SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK3ab7192c;rport
From: "asterisk" <sip:asterisk@xxx>;tag=as02cd9ac9
To: <sip:201@10.1.50.110>
Contact: <sip:asterisk@xxx>
Call-ID: 5750e0ba4bcdade11355d72d32d79ae8@xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 08 Jan 2007 06:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Transmitting (no NAT) to 10.1.50.110:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.50.110;branch=z9hG4bKeb17b034FEA2504;received=10.1.50.110
From: "201" <sip:201@xxx>;tag=5F13B47-642AE67A
To: <sip:201@xxx>
Call-ID: 17f4ccb-2adb0721-fbc9ee6c@10.1.50.110
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:201@xxx>
Content-Length: 0


---
Transmitting (no NAT) to 10.1.50.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.50.110;branch=z9hG4bKeb17b034FEA2504;received=10.1.50.110
From: "201" <sip:201@xxx>;tag=5F13B47-642AE67A
To: <sip:201@xxx>;tag=as3dd7fe57
Call-ID: 17f4ccb-2adb0721-fbc9ee6c@10.1.50.110
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 3600
Contact: <sip:201@10.1.50.110>;expires=3600
Date: Mon, 08 Jan 2007 06:47:23 GMT
Content-Length: 0


---
Destroying call '5750e0ba4bcdade11355d72d32d79ae8@xxx'
Scheduling destruction of call '17f4ccb-2adb0721-fbc9ee6c@10.1.50.110' in 15000 ms
    -- SIP Seeding peer from astdb: '201' at 201@10.1.50.110:5060 for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.50.110:5060:
OPTIONS sip:201@10.1.50.110 SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK35b72266;rport
From: "asterisk" <sip:asterisk@xxx>;tag=as08e4e1c6
To: <sip:201@10.1.50.110>
Contact: <sip:asterisk@xxx>
Call-ID: 3cd61ef74bb84bb90c90cd793956ba28@xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 08 Jan 2007 06:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '3cd61ef74bb84bb90c90cd793956ba28@xxx'
tel*CLI>
<-- SIP read from 10.1.50.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK3ab7192c;rport
From: "asterisk" <sip:asterisk@xxx>;tag=as02cd9ac9
To: <sip:201@10.1.50.110>;tag=52F327FF-E9C30A52
CSeq: 102 OPTIONS
Call-ID: 5750e0ba4bcdade11355d72d32d79ae8@xxx
Contact: <sip:201@10.1.50.110>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Length: 0


--- (10 headers 0 lines)---
Destroying call '5750e0ba4bcdade11355d72d32d79ae8@xxx'
tel*CLI>
<-- SIP read from 10.1.50.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK35b72266;rport
From: "asterisk" <sip:asterisk@xxx>;tag=as08e4e1c6
To: <sip:201@10.1.50.110>;tag=D527049E-B22117B5
CSeq: 102 OPTIONS
Call-ID: 3cd61ef74bb84bb90c90cd793956ba28@xxx
Contact: <sip:201@10.1.50.110>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067
Content-Length: 0

thanks for the help!

6

hmmm i dont see a notify for mwi in that… i think your problem is on the * side. does using another phone work?

also maybe post your voicemail.conf and the sip.conf entry?

7

voicemail (in realtime)

customer_id = 201
context = default
mailbox = 201
fullname = xxx
email = xxx
pager = 
stamp = NULL
password = xxx

sip (in realtime)

name = 201
callerid = xxx
canreinvite = no
context = internal
dtmfmode = NULL
host = dynamic
mailbox = 201@default
nat = no
port = 5060
qualify = yes
secret = xxx
type = friend
username = 201
disallow = all
allow = g726;ilbc;gsm;ulaw;alaw
musiconhold = default
regseconds = 1168312638
ipaddr = 10.1.50.110
cancallforward = yes

anything look wrong?

8

you’re using realtime then? try turning rtcache friends on in sip.conf
i’m pretty sure that was my problem with my polycom 301’s

9

rtcache fixed that problem, now however I am unable to communicate with a ATA behind a nat, it keeps retransmitting over and over. Any ideas?

Also I read that with rtcache the server does not refresh the database like it should(ie remove old users), Is there any way to make the cache refresh every so often?

10

if you reconnect to the asterisk console (asterisk -r) and issue a ‘reload’ statement, it will reload all config and should pull out old realtime users. I guess you could do that with a cron job, ‘asterisk -rx reload’ should do the job nicely :wink: