[asterisk-users] SIP Presence not working

[Deka, Rajib IN MAA SL]

Hi All,

Following message I got in console for an extension,

[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c:
<--- SIP read from UDP:132.186.230.70:7510 --->
SUBSCRIBE sip:18215 at sip1.test.in SIP/2.0^M
Via: SIP/2.0/UDP 132.186.230.70:7510;branch=z9hG4bK-d8754z-2b3b65532b3b6553-1---d8754z-;rport^M
Max-Forwards: 70^M
Contact: <sip:18238 at 132.186.230.70:7510>^M
To: <sip:18215 at sip1.test.in>^M
From: "18238"<sip:18238 at 10.20.20.52>;tag=2b3b6553^M
Call-ID: MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.^M
CSeq: 1 SUBSCRIBE^M
Subject: Available^M
Expires: 3600^M
Accept: multipart/related, application/rlmi+xml, application/pidf+xml^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO^M
Supported: replaces^M
User-Agent: ^M
Event: presence^M
Content-Length: 0^M
^M

<------------->
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: --- (16 headers 0 lines) ---
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Ignoring this SUBSCRIBE request
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Found peer '18238' for '18238' from 132.186.230.70:7510
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Looking for 18227 in test-local-outgoing (domain sip1.test.in)
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Scheduling destruction of SIP dialog 'MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.' in 3610000 ms (Method: SUBSCRIBE)
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c:
<--- Transmitting (no NAT) to 132.186.230.70:7510 --->
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 132.186.230.70:7510;branch=z9hG4bK-d8754z-2b3b65532b3b6553-1---d8754z-;received=132.186.230.70;rport=7510^M
From: "18238"<sip:18238 at 10.20.20.52>;tag=2b3b6553^M
To: <sip:18215 at sip1.test.in>;tag=as6c37f730^M
Call-ID: MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.^M
CSeq: 1 SUBSCRIBE^M
Server: Asterisk PBX 1.6.2.16.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M
Supported: replaces, timer^M
Expires: 3600^M
Contact: <sip:18227 at 10.20.20.52>;expires=3600^M
Content-Length: 0^M

<------------>
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: set_destination: Parsing <sip:18238 at 132.186.230.70:7510> for address/port to send to
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: set_destination: set destination to 132.186.230.70, port 7510
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Reliably Transmitting (no NAT) to 132.186.230.70:7510:
NOTIFY sip:18238 at 132.186.230.70:7510 SIP/2.0^M
Via: SIP/2.0/UDP 10.20.20.52:5060;branch=z9hG4bK0f6d2ae2;rport^M
Max-Forwards: 70^M
From: <sip:18227 at sip1.siemens.in>;tag=as6c37f730^M
To: "18238"<sip:18238 at 10.20.20.52>;tag=2b3b6553^M
Contact: <sip:18227 at 10.20.20.52>^M
Call-ID: MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.^M
CSeq: 119 NOTIFY^M
User-Agent: Asterisk PBX 1.6.2.16.1^M
Event: presence^M
Content-Type: application/pidf+xml^M
Subscription-State: active^M
Content-Length: 533^M
^M
<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:18238 at 10.20.20.52">
<pp:person><status>
<ep:activities><ep:away/></ep:activities>
</status></pp:person>
<note>Not online</note>
<tuple id="18227">
<contact priority="1">sip:18227 at sip1.siemens.in</contact>
<status><basic>closed</basic></status>
</tuple>
</presence>


________________________________
From: Deka, Rajib IN MAA SL
Sent: Tuesday, July 05, 2011 12:15 PM
To: 'asterisk-users at lists.digium.com'
Subject: SIP Presence not working

Hello all,

I found a problem regarding SIP presence in asterisk (1.6.2). The scenario is not working properly for all users.
Our SIP client sends SIP:SUBSCRIBE to all the configured extensions in asterisk during registration process. Asterisk replies with 200 OK for all SUBSCRIBE. But if I run "sip show subscriptions" in CLI prompt, it shows only a few live subscriptions per user. The result is not consistent; sometime it shows subscription status for all the extensions and sometime a few (per user). We have allowsubscribe=yes and callcounter=yes in sip.conf file.

Can somebody please help me to debug this issue and identify the root cause?

Regards,
Rajib

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