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Thread: discussion Digest, Vol 27, Issue 5
discussion Digest, Vol 27, Issue 5
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United States		 2007-10-03 08:36:06 
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Today's Topics:

   1. 302 message question (Mihaly Zachar)
   2. Re: 302 message question (sreekant nair)
   3. (no subject) (Amos Halfon)
   4. Call Disconnect issue with Cisco AS5300	running SIP
(Louis Wu)
   5. FMTP / RTPMAP (Mhike)
   6. Re: FMTP / RTPMAP (Abhishek Mishra)
   7. CISCO 7940 TFTP Timeout (Ian Sivell)
   8. TekSIP (Yasin KAPLAN)
   9. Image with invite msg (amit)


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

Message: 1
Date: Tue, 02 Oct 2007 18:07:10 +0200
From: Mihaly Zachar <zmihalymadein.hu>
Subject: [SIPForum-discussion] 302 message question
To: discussionsipforum.org
Message-ID: <47026CAE.6030603madein.hu>
Content-Type: text/plain; charset=ISO-8859-2; format=flowed

Hi all,

I'm writing an UAS.

The UAS has a feature, that if the called number is matching
with a 
pattern, it will send 183 Session in progress, than plays an
RTP stream 
and then redirect the UAC with 302 Moved Temporarily..

There is an UAC, and it's developers says that I should not
send 302 
Redirect after the 183 Session in progress.

This solution works well with CISCO media gateways.

I can't find it in the RFC 3261 who has the truth..

Can sy help me in this ?


So, the call flow is the following:

UAC                      UAS
      --- INVITE --->
      <--- 100 ------
      <--- 183 ------
      <-- RTP --
           .
           .
           .
      <--- 302 ----


Is this correct ?


Thanks,
Misi


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

Message: 2
Date: Tue, 2 Oct 2007 11:05:42 -0700 (PDT)
From: sreekant nair <sreekant_nairyahoo.com>
Subject: Re: [SIPForum-discussion] 302 message question
To: Mihaly Zachar <zmihalymadein.hu>,
discussionsipforum.org
Message-ID: <11724.69746.qmweb51107.mail.re2.yahoo.com>
Content-Type: text/plain; charset="us-ascii"

Check out this link. 
http://www.cisco.co
m/univercd/cc/td/doc/product/software/ios122/rel_docs/sip_fl
o/hennigan.htm

There is a call flow depicting the messaging for a scenario
where a 3XX response is received after a 183 is sent by the
server. I guess that explains how CISCO supports it. But
yeah I need to dig deeper to find an RFC that states this. 

Regards
Sreekant

----- Original Message ----
From: Mihaly Zachar <zmihalymadein.hu>
To: discussionsipforum.org
Sent: Tuesday, October 2, 2007 12:07:10 PM
Subject: [SIPForum-discussion] 302 message question

Hi all,

I'm writing an UAS.

The UAS has a feature, that if the called number is matching
with a 
pattern, it will send 183 Session in progress, than plays an
RTP stream 
and then redirect the UAC with 302 Moved Temporarily..

There is an UAC, and it's developers says that I should not
send 302 
Redirect after the 183 Session in progress.

This solution works well with CISCO media gateways.

I can't find it in the RFC 3261 who has the truth..

Can sy help me in this ?


So, the call flow is the following:

UAC                      UAS
      --- INVITE --->
      <--- 100 ------
      <--- 183 ------
      <-- RTP --
           .
           .
           .
      <--- 302 ----


Is this correct ?


Thanks,
Misi
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Message: 3
Date: Wed, 3 Oct 2007 07:49:20 +0200
From: "Amos Halfon" <amos.halfongmail.com>
Subject: [SIPForum-discussion] (no subject)
To: discussionsipforum.org
Message-ID:
	<697963e10710022249o2bd13e6ayd95a4d6c12a97794mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"


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Message: 4
Date: Wed, 3 Oct 2007 14:11:58 +0800
From: "Louis Wu" <louisttv.com.hk>
Subject: [SIPForum-discussion] Call Disconnect issue with
Cisco AS5300
	running SIP
To: discussionsipforum.org
Message-ID: <20071003055430.M56996ttv.com.hk>
Content-Type: text/plain;	charset=gb2312

Hi All,

I have a Cisco AS5300 using SIP and initiate SIP call to a
SIP server YATE (v 
1.3.0). 

I have a call disconnect problem whenever my Cisco receive a
183 Session 
Progress message from the YATE server. The symptoms are
listed as below.

1. Cisco AS5300 send an INVITE to the YATE server
2. YATE returns a 100 Trying message
3. YATE returns a 183 Session Progress
4. YATE returns a 200 OK 
5. Two-way-audio starts (start conversation as usual), but
at the ISDN side 
of the Cisco, the call is shown to be "not
connected"
6. Cisco sends a ACK
7. Cisco sends a BYE
8. YATE returns a 100 Trying
9. Cisco sends a BYE
10. YATE returns a 200 OK
11. Call disconnect with status message saying "no
answer" at the calling 
party's mobile handset
12. Cisco logs a Disconnect Cause (CC) : 16 (SIP) : 200

If the YATE returns a 180 Session Progress in (3) above, the
call will be 
connected normally and works as usual.

Please give me your professional advice and resolution on
the above 
disconnect issue.

Cheers
Louis



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

Message: 5
Date: Wed, 3 Oct 2007 01:30:41 -0700 (PDT)
From: Mhike <mhiqeyahoo.com>
Subject: [SIPForum-discussion] FMTP / RTPMAP
To: discussionsipforum.org
Message-ID: <486570.72622.qmweb50811.mail.re2.yahoo.com>
Content-Type: text/plain; charset="us-ascii"

Hi,

Can anyone tell me what's the meaning of FMTP and RTPMAP?
What are their differences?

Thanks.



     
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Message: 6
Date: Wed, 03 Oct 2007 14:17:18 +0530
From: Abhishek Mishra <abhishek.mishragloballogic.com>
Subject: Re: [SIPForum-discussion] FMTP / RTPMAP
To: Mhike <mhiqeyahoo.com>
Cc: discussionsipforum.org
Message-ID: <1191401237.2587.6.camellinux.site>
Content-Type: text/plain

Hi Mhike,

Please refer to RFC 2327 and RFC 3264:
http://tools.ietf.
org/html/rfc2327

Kind Regards,
-Abhishek

On Wed, 2007-10-03 at 14:00, Mhike wrote:
> Hi,
> 
> Can anyone tell me what's the meaning of FMTP and
RTPMAP? What are
> their differences?
> 
> Thanks.
> 
> 
>
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------------------------------

Message: 7
Date: Wed, 3 Oct 2007 12:36:29 +0100
From: "Ian Sivell" <ian.sivellgmail.com>
Subject: [SIPForum-discussion] CISCO 7940 TFTP Timeout
To: discussionsipforum.org
Message-ID:
	<123aa00e0710030436y6996fd33o577aed42e77b49fmail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

I hope someone can help me

I have recently bought a used 7940 from EBay, initially it
booted and I
could get into the menus (all be them locked), I found on
the cisco site to
hold down th # key whilst powering up and then enter
123456789*0# to reset
toi factory defaults.

Since doing is the phone boots but just stays at the tftp
timeout message.

On the Cisco site it says that the phone should timeout
after 20 seconds and
continue to boot correctly after that giving access to the
menus mine does
not seem to do this.

It has been loaded with SCCP (Skinny) as far as I can tell,
and the DHCP
server on my network is assigning it a DHCP address within a
valid subnet
but still it times out and does not get any further than the
message above.

I have reset this several times to n avail

Please some one help me

Thanks
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Message: 8
Date: Wed, 3 Oct 2007 14:46:47 +0300
From: "Yasin KAPLAN" <yasinkaplan.net>
Subject: [SIPForum-discussion] TekSIP
To: <discussionsipforum.org>
Message-ID: <015001c805b3$163e84e0$1a0d3ad4doruk.com.tr>
Content-Type: text/plain; charset="us-ascii"

Hi,

 

I've recently released beta version of TekSIP Registrar
& Proxy:

 

http://www.teksip.com/

 

You feedback is welcomed.

 

Thanks,

 

Yasin KAPLAN

 

 

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Message: 9
Date: Wed, 03 Oct 2007 19:16:03 +0530
From: amit <amit.vpyronetworks.com>
Subject: [SIPForum-discussion] Image with invite msg
To: discussionsipforum.org
Message-ID: <1191419164.5119.3.camelamit>
Content-Type: text/plain

Hi All,



	How we send image with sip invite message ?


Thanks in Advance

Amit  






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

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