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Today's Topics:
1. Re: difference between INFO and OPTIONS method? (Halit
Sakca)
2. Re: Are "<" and ">"
mandatory? (Halit Sakca)
3. Multiple DTMF events received for a single DTMF event
(Konstantin Bokarius)
4. Re: Multiple DTMF events received for a single DTMF
event
(Konstantin Bokarius)
5. Re: Multiple DTMF events received for a single DTMF
event
(Konstantin Bokarius)
6. Re: is there a Timer for 200 OK? (Herve Jourdain)
------------------------------------------------------------
----------
Message: 1
Date: Sun, 9 Dec 2007 19:09:27 +0200
From: Halit Sakca <sakcahalithotmail.com>
Subject: Re: [SIPForum-discussion] difference between INFO
and OPTIONS
method?
To: nidhi jain <nidhi_jain6680yahoo.co.in>,
<discussionsipforum.org>
Message-ID: <BAY111-W5450FF92E1EB052BCF18CCC6A0phx.gbl>
Content-Type: text/plain; charset="windows-1254"
Hey,Q: How are DTMF digits sent before the session setup?
A: You should configure your mediagateway also residential
gw according to your network. (AS or Proxy)
Halit
Date: Fri, 7 Dec 2007 15:09:33 +0000From: nidhi_jain6680yahoo.co.inTo: discussionsipforum.orgSubject:
[SIPForum-discussion] difference between INFO and OPTIONS
method?Hello, CANCEL request is sent when response
to INVITE message is not received. If this INVITE request is
lost and UA may still send a CANCEL request. Is this
possible? Can CANCEL request be sent without sending an
INVITE message? What is the basic difference between
INFO and OPTIONS method? can OPTIONS used for mid-call
signalling? Why is ACK request required? Is it just
to avoid retransmission of response message. This could be
done even by starting the media transmission. How
are DTMF digits sent before the session setup? Please reply.
Thank you.
Get the freedom to save as many mails as you wish. Click
here to know how.
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Message: 2
Date: Sun, 9 Dec 2007 19:14:18 +0200
From: Halit Sakca <sakcahalithotmail.com>
Subject: Re: [SIPForum-discussion] Are "<" and
">" mandatory?
To: Andrea Puddu <androjokerhotmail.com>,
<discussionsipforum.org>
Message-ID: <BAY111-W2365AFF0240D45A4D6A403CC6A0phx.gbl>
Content-Type: text/plain; charset="windows-1254"
you are welcome, if you cant take trace try ethereal then.
halit,
From: androjokerhotmail.comTo: sakcahalithotmail.comSubject: RE: [SIPForum-discussion] Are
"<" and ">" mandatory?Date: Fri, 7
Dec 2007 08:47:09 +0000
Thanks for your feedback. It is not so easy to have logs
from SBC and AS :(Anyway I keep on investigating.It seems
weird to me that SBC sends only a BYE in answer to the REFER
request and not for example a 400 Bad Request.Andrea
From: sakcahalithotmail.comTo: androjokerhotmail.comSubject: RE: [SIPForum-discussion] Are
"<" and ">" mandatory?Date: Thu, 6
Dec 2007 22:28:51 +0200
hey Andrea,<> is not mandatory, 202 should be sent by
AS so focus on that. or sbc could block it, is it possible
to have logging from both AS, sbc?regards,halit
From: androjokerhotmail.comTo: discussionsipforum.orgDate: Wed, 5 Dec 2007 18:59:23
+0000Subject: [SIPForum-discussion] Are "<" and
">" mandatory?
Hello,I am investigating on an issue regarding the call
transfer scenario.By now, the main issue seems to be the
missing response by the Application Server to the NOTIFY
request of the transferor party.First of all, after the
NOTIFY request, I expect the SBC (or the Application Server)
to send a 202 Accepted. This does not happen. The SBC does
not send anything.Having a look at the NOTIFY syntax, I
noticed that "<" and ">" are
missing in some filed, like "Refer-to" and
"Referred-by".Does anyone know if "<"
and ">" are mandatory to delimit all SIP header
fields?Thanks 1000,Andrea
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t?kla!
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Message: 3
Date: Sun, 9 Dec 2007 15:29:31 -0800
From: "Konstantin Bokarius" <bokariuscomcast.net>
Subject: [SIPForum-discussion] Multiple DTMF events received
for a
single DTMF event
To: <discussionsipforum.org>
Message-ID: <000001c83abb$59f67ac0$0de37040$net>
Content-Type: text/plain; charset="us-ascii"
I have a call bridged through Asterisk to another SIP
system. Unless the
't' option in the Dial application is specified (this option
is to allow the
called user to redirect the callee) each time a DTMF event
is received by
Asterisk from the callee the called SIP system received 4-6
multiples of
that DTMF event. So if the callee hit '16' on his keypad
the receiving
system might receive '1111666666'.
When Asterisk dials another kind of SIP system I don't see
this problem -
only when dialing a specific system (and that system's DTMF
features cannot
be configured).
Has anyone experienced anything like this?
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Message: 4
Date: Sun, 9 Dec 2007 15:35:11 -0800
From: "Konstantin Bokarius" <bokariuscomcast.net>
Subject: Re: [SIPForum-discussion] Multiple DTMF events
received for a
single DTMF event
To: <discussionsipforum.org>
Message-ID: <000b01c83abc$250ab320$6f201960$net>
Content-Type: text/plain; charset="us-ascii"
This issue happens with all current versions of Asterisk.
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Message: 5
Date: Sun, 9 Dec 2007 16:41:52 -0800
From: "Konstantin Bokarius" <bokariuscomcast.net>
Subject: Re: [SIPForum-discussion] Multiple DTMF events
received for a
single DTMF event
To: <discussionsipforum.org>
Message-ID: <001901c83ac5$759e3510$60da9f30$net>
Content-Type: text/plain; charset="us-ascii"
Inband prevents DTMF events from getting through at all. I
have tried all
the DTMF modes and the only one that works is RFC2833 and it
is the one that
is having the issue.
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Message: 6
Date: Mon, 10 Dec 2007 07:50:54 +0100
From: "Herve Jourdain" <herve.jourdainmstarsemi.com>
Subject: Re: [SIPForum-discussion] is there a Timer for 200
OK?
To: =?gb2312?B?J9PqILPCJw==?= <chen.yu26yahoo.com.cn>,
<discussionsipforum.org>
Message-ID: <200712100646.lBA6k5QK052116mailsqr.mstarsemi.com>
Content-Type: text/plain; charset="gb2312"
Hi,
I think you mean a timer when UAS sent a 200 OK response,
and waiting for
ACK from UAC, is that right?
If it??s so, as far as I know, there is no such timer at the
State Machine
level.
One of the reasons is that the ACK will have a different via
branch, and
hence will not be matched to the previous transaction (it??s
not the case
for other ??error?? responses, which are dealt with by the
State Machine).
Nevertheless, RFC 3261 states (13.3.1.4, p.85), that the 2xx
response should
be periodically passed by the UAS core to the transport ?C
and not using the
state machine ?C until the ACK arrives.
The retransmission interval starts at T1, and doubles until
it reaches T2
(basically like non-invite transactions).
At the implementation level, though, you could consider
??tricking?? the
state machine to do the retransmission, provided you have a
way when
receiving the ACK to link it with this state machine, and
remove it. It
really depends on your implementation there.
Regards,
Herv??
_____
From: discussion-bouncessipforum.org
[mailto:discussion-bouncessipforum.org] On Behalf Of
? ?
Sent: samedi 8 d??cembre 2007 10:32
To: discussionsipforum.org
Subject: [SIPForum-discussion] is there a Timer for 200 OK?
Hi,everyone in the forum!
I have a problem and want U to give me a hand
that is:
Is there a special timer started when UAS received a 200 OK
response, and
this timer will fire if the UAS can't get ACK back from UAC
in a period?
I found the RFC3621 just mentioned a Timer H which will
start for 3**-6**
responses.
thanks
nora
_____
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x.html?source=xy> ??????????
??????????????????????????
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