Buddy Presence - Polycom IP 550 - Asterisk 1.4.17

Asterisk Asterisk Support
1

Dear * users,

I am working on a deployment project and need to know all the points to fix and to configure in order to get two phones able to see the state of the other. The problem is it is quite urgent as deployment is delayed for this only.

To resume, a secretary phone ext 3000 and a director phone ext 3001.

Secretary should be able to see the state of the line of the director before forwarding calls to him (having a call or not will be enough!).

I looked all over the Internet and found parts of information but nothing works. I use Asterisk 1.4.17 and Polycom IP Phones 550.

What are the different steps I need to follow in order to get all this working fine?

Best regards.

2

Off the top of my head, I think all you need to do is set limitonpeers=yes in sip.conf,
set up the hints in extensions.conf,
exten => 3001,hint,SIP/3001
exten => 3000,hint,SIP/3000

and set up the -directory.xml files to enable buddy watch for the extension in the directory 1.

<?xml version="1.0" encoding="UTF-8" standalone="yes"?> <directory> <item_list> <item> <fn>Director First Name</fn> ;first name <ln>Director Last Name</ln> ;last name <ct>3001</ct> ;actual extension <sd>1</sd> ;speed-dial index <rt>6</rt> ;ring type <dc/> ;divert contact - usually blank <ad>0</ad> ;auto divert <ar>0</ar> ;auto reject <bw>1</bw> ;enables buddy watching - used to set up presence <bb>0</bb> ;buddy block - blocks this contact from watching this phone </item> </item_list> </directory>

3

Thank you very much for your reply !

Concerning this, where do I need to setup those hints ? only in the context that phones are in ? Before the Dial application ? or somewhere else ?

Thanks in advance & best regards !

4

You’d set them up in the context the phones are in. I also forgot, you need to set a call-limit on the sip peers as well.

5

Thank you very much, I will try to apply the changes tomorrow,

I already did the work for the in both mac-directory.xml & call-limit & limiteonpeers but the hint in the dialplan looked doing nothing when call is going over it.

Best regards,

6

I can see the Buddies tab but the phone I want to monitor is offline (same on other phone)

When I make a call, nothing changes on the other phone in the Buddies tab.

This is my configuration :

sip.conf :

[code][2069]
type=friend
context=be
username=2069
callerid=***
secret=***
host=dynamic
callgroup=6
pickupgroup=6
dtmfmode=rfc2833
qualify=yes
call-limit=1
limitonpeers=yes

[2056]
type=friend
context=be
username=2056
qualify=yes
md5secret=***
callerid=***
pickupgroup=6
callgroup=6
host=dynamic
dtmfmode=rfc2833
call-limit=1
limitonpeers=yes[/code]

extensions.conf :

[code];TEST BUDDY 2056
exten => 2056,hint,SIP/2056
exten => 2056,1,Dial(SIP/${EXTEN},180,rt)
exten => 2056,2,Hangup

;TEST BUDDY
exten => 2069,hint,SIP/2069
exten => 2069,1,Dial(SIP/${EXTEN},180,rt)
exten => 2069,n,Hangup[/code]

in each mac-directory.xml :

[code]
***
***
2056
1
1

            <item>
                    <ln>***</ln>
                    <fn>***</fn>
                    <ct>2069</ct>
                    <bw>1</bw>
                    <sd>1</sd>
            </item>[/code]

How can I fix all this?

Another thing, when I try to send a message to the other phone, 2069 to 2056 using the Buddies msg sender, I receive this on Asterisk :

Thanks in advance !

7

limitonpeers should be in the general section of sip.conf not on the peer definition.

8

I have just upgraded to Asterisk 1.6.0.3-rc1.

When registering my Ekiga client 3.0.1 and change my presence status, I receive the message below ‘SIP/2.0 501 Method Not Implemented’.

I have followed the previous instructions in your post.

Do you have any suggestions what I might be doing wrong?

linux-2uvo*CLI>
<— SIP read from UDP://192.168.4.57:5060 —>
PUBLISH sip:1002@192.168.4.102 SIP/2.0
Route: sip:192.168.4.102:5060;lr
CSeq: 4 PUBLISH
Via: SIP/2.0/UDP 192.168.4.57:5060;branch=z9hG4bK74db47cf-31fd-1810-9ba4-001d0940241d;rport
User-Agent: Ekiga/3.0.1
From: sip:1002@192.168.4.102;tag=10db47cf-31fd-1810-9ba4-001d0940241d
Call-ID: 10db47cf-31fd-1810-9ba3-001d0940241d@CPKS1682
To: sip:1002@192.168.4.102
Contact: sip:1002@192.168.4.57
Expires: 300
Event: presence
Content-Type: application/pidf+xml
Content-Length: 306
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?> away open sip:1002@192.168.4.102

<------------->
— (14 headers 10 lines) —
linux-2uvo*CLI>
<— Transmitting (NAT) to 192.168.4.57:5060 —>
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 192.168.4.57:5060;branch=z9hG4bK74db47cf-31fd-1810-9ba4-001d0940241d;received=192.168.4.57;rport=5060
From: sip:1002@192.168.4.102;tag=10db47cf-31fd-1810-9ba4-001d0940241d
To: sip:1002@192.168.4.102;tag=as657fc7f5
Call-ID: 10db47cf-31fd-1810-9ba3-001d0940241d@CPKS1682
CSeq: 4 PUBLISH
User-Agent: Asterisk PBX 1.6.0.3-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0