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Thread: discussion Digest, Vol 26, Issue 5
discussion Digest, Vol 26, Issue 5
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United States		 2007-09-03 05:30:24 
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Today's Topics:

   1. voicemail to presence interface (Sai P. Varanasi)
   2. how to find outgoing calls (Renjith Mamman (NeSTIT))
   3. Any opensource Presence server around (Saurabh
Agarwal)
   4. Which one maintain the route information in	SIP,
transaction
      layer or IP layer? (mark lee)
   5. Re: Any opensource Presence server around (Asha
Pillai)
   6. Query regarding RTP/AVP Media Profiles (Matthew
Pragnell)


------------------------------------------------------------
----------

Message: 1
Date: Mon, 3 Sep 2007 14:23:05 +0530
From: "Sai P. Varanasi" <sai.varanasixius.com>
Subject: [SIPForum-discussion] voicemail to presence
interface
To: <discussionsipforum.org>
Message-ID: <200709030838.l838c9Zb020135serv1.xius.com>
Content-Type: text/plain; charset="us-ascii"

Hi All,

 

  Is there a standard for a third party voicemail subsystem
to notify the
presence server that a user has a new message? How does the
presence server
notify a particular user about MWI on a voicemail engine?

 

Thanks & Regards,

Sai Prabhakar Varanasi

 

 

Thanks & Regards,


Sai Prabhakar Varanasi

Module Lead, R & D Team,

Ph: +91 (40) 44330022 Xtn: 6011

  

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Message: 2
Date: Mon, 3 Sep 2007 14:23:42 +0530
From: "Renjith Mamman (NeSTIT)"
<renjith.mammannestgroup.net>
Subject: [SIPForum-discussion] how to find outgoing calls
To: <govindraj_hyahoo.co.in.com>, "Vijay T"
<vijaycecyahoo.co.in>,
	<discussionsipforum.org>
Message-ID:
	<DFA4DD36883F5641AD609D011918C1B33A78F5new-ex1.nestgroup.net>
Content-Type: text/plain; charset="iso-8859-1"

dear all,
    how can i find outgiong/incoming calls from the cdrs
(call detail record) . 
 
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Message: 3
Date: Mon, 3 Sep 2007 14:38:56 +0530
From: "Saurabh Agarwal" <sagarwal1981gmail.com>
Subject: [SIPForum-discussion] Any opensource Presence
server around
To: discussionsipforum.org
Message-ID:
	<faf00aaa0709030208v14a932c0g312c7228e5b03ccmail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi All,

Anybody aware of any opensource Presence server (IMS
perspective ).

Please point me to some good documents pointing to
requirments etc of
Presence server.


-- 
Thanks
Saurabh Agarwal
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Message: 4
Date: Mon, 3 Sep 2007 18:02:15 +0800 (CST)
From: mark lee <gogofly_leeyahoo.com.cn>
Subject: [SIPForum-discussion] Which one maintain the route
	information in	SIP, transaction layer or IP layer?
To: sip sip <discussionsipforum.org>
Message-ID: <949415.88781.qmweb15904.mail.cnb.yahoo.com>
Content-Type: text/plain; charset="gb2312"

Hello, everyone:
  As you know, SIP defines strict route and loose route,
could you tell me which layer should maintain the route
information in couple of cases, transaction layer or IP
layer?
   
  Thanks a lot
   
  Mark

       
---------------------------------
???????????????????????????????????????????? 
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Message: 5
Date: Mon, 3 Sep 2007 15:44:48 +0530
From: "Asha Pillai" <asha.g.pillaigmail.com>
Subject: Re: [SIPForum-discussion] Any opensource Presence
server
	around
To: discussionsipforum.org
Message-ID:
	<2024689c0709030314v57402dc7re1d3a8ea447d65bemail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi

I tried to run the resiprocate SIP code in visual studio and
it was diving
an error that cppunit.vcproj was not there for tfm
solution.Can anybody help
in this regard
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Message: 6
Date: Mon, 3 Sep 2007 11:29:58 +0100
From: "Matthew Pragnell" <Matthew.Pragnelltnzi.com>
Subject: [SIPForum-discussion] Query regarding RTP/AVP Media
Profiles
To: discussionsipforum.org
Message-ID: <0E0491FBA51DD5418554BEA5A1E12D52C82F97ln1.tnzi.com>
Content-Type: text/plain; charset="us-ascii"

Hi,

 

I wonder if anyone can help me. 

 

I have a fault which is affecting our IP Call Control
platform. I
receive an INVITE, in the SDP, the codec list specifies g729
variants
(Media Description, name and address (m): audio 32206
RTP/AVP 18 18 18
18 0 8 96). I think that the reason it is hanging in our IP
call element
is because there is some missing media attribute information
in the SDP
for example (a=fmtp:18 annexb=no). This specifies which
variant of G729
to use. 

 

SDP

v=0

o=msx-mitry-2 2147483647 1 IN IP4 x.x.x.x

s=sip call

c=IN IP4 84.103.233.85

t=0 0

m=audio 35864 RTP/AVP 18 18 18 18 0 8 96

a=rtpmap:96 telephone-event/8000

a=sendrecv

a=ptime:20

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729B/8000

a=rtpmap:18 G729AB/8000

a=rtpmap:18 G729A/8000

a=rtpmap:18 G729/8000

m=image 35866 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPRedundancy

a=T38MaxBitRate:14400

a=T38FaxVersion:0

 

Can someone confirm if this is correct and add any
additional
information which would be helpful.

 

Thanks

 

Matt Pragnell
UK Network Support Engineer 

T

+44 207 628 9833

 

F

+44 207 628 5262

M

+44 77 9542 6125

E

matthew.pragnelltnzi.com <mailto:steve.burgesstnzi.com> 

W

www.tnzi.com <http://www.tnzi.com/>


TNZUK Suite, C/O IX Europe 
101, Finsbury Pavement, 
bond, EC2A 1RS, UK

 


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