It's a deregister request. So you need to remove all the bindings for that AoR.
-Original Message-
From: brezden [mailto:brez...@gmail.com]
Sent: Wednesday, May 08, 2013 10:22 AM
To: Keerthi Srinivasan
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] REGISTER mes
If the voicemail (VM) is present for Party B, then call will be answered
by VM once predefined time is expired.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Aman
Sent: Tuesday, May 22, 2012 4:42
UAC must send ACK with SDP (specifying that media (audio/video/image)
port to 0).
For details please refer to:
http://tools.ietf.org/rfc/rfc4317.txt
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
That's what I suggested. It makes logical sense.
Thanks,
-Original Message-
From: Vivek Talwar [mailto:vivek.tal...@frogdesign.com]
Sent: Friday, August 26, 2011 2:46 AM
To: Worley, Dale R (Dale); Uttam Sarkar (usarkar); prakash k;
sip-implementors@lists.cs.columbia.edu
Subject: RE:
Looks like it's a conflict request from a UAC. One can have multiple
Privacy header in SIP message with value specified as "user", "session",
"header", "critical". When it specifies "none" then other values are
meaningless.
I think you can choose behavior of "none". That's my 2 cents.
Thanks,
I agree with Inaki. We did exactly the same.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Romel Khan
Sent: Friday, August 12, 2011 7:05 PM
To: Iñaki Baz Castillo
Cc: sip-implementors@lists.cs.c
Here are 2 RFC that may help you.
IPv6 Transition in SIP:
http://tools.ietf.org/html/rfc6157
SIP:
http://tools.ietf.org/html/rfc3261
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
guan xsun
S
Try this.
From: Siga [mailto:fruchta...@googlemail.com]
Sent: Monday, March 28, 2011 9:48 AM
To: Uttam Sarkar (usarkar)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] RTP Send thread problem
Hi Uttam,
I did the changes as per your request (tried both the cases
Please see inline.
From: Siga [mailto:fruchta...@googlemail.com]
Sent: Monday, March 28, 2011 8:00 AM
To: Uttam Sarkar (usarkar)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] RTP Send thread problem
Hi Uttam,
about the variable for the sequence number I using
Make sure you use static or global variable for sequence number in
rtp_send_packets.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Siga
Sent: Monday, March 28, 2011 6:53 AM
To: sip-implementors@l
If the transport is UDP then UA must retransmit request again if it does
not receive any response within T1 time period. Default value of T1 is
500 ms.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf
4341ae6cbe5a359", opaque="",
uri="sips:ss2.biloxi.example.com",
-à
From: Siga [mailto:fruchta...@googlemail.com]
Sent: Tuesday, March 22, 2011 9:12 AM
To: Uttam Sarkar (usarkar)
Cc: sip-implementors@lists.cs.columbia.
Please check you X-Lite configuration. It must have been configure with
required authentication.
So, your client must send INVITE with credentials to overcome 401
response.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.col
-Original Message-
From: Sambasiva Rao Manchili
[mailto:sambasiva.manch...@nexustelecom.com]
Sent: Monday, March 21, 2011 11:37 AM
To: Uttam Sarkar (usarkar); Sambasiva Rao MANCHILI;
sip-implementors@lists.cs.columbia.edu
Cc: Antonio Gambin
Subject: RE: [Sip-implementors] SIP UDP packet
There could be bottle neck in your application. Maybe it's unable to
read all the UDP packets from the network.
You need to find out what is the capacity of your application.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.c
Looks like your customer is sending "hostname" incorrectly. Probably it
could have been your proxy/B2BUA/SBC's IP address of FQDN. Then your
SBC could have resolve that FQDN or hostname to forward the message to
that particular proxy/B2BUA to handle call.
As, your SBC is trying to resolve "none" a
That's right.
-Original Message-
From: Bob Penfield [mailto:bpenfi...@acmepacket.com]
Sent: Monday, November 29, 2010 10:24 AM
To: Uttam Sarkar (usarkar); varun;
sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] dynamic payload negotiation
You have it backwards
Correction..
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Uttam Sarkar (usarkar)
Sent: Monday, November 29, 2010 9:54 AM
To: varun; sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip
Please see inline.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
varun
Sent: Monday, November 29, 2010 5:31 AM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] dynamic paylo
PM
To: Uttam Sarkar (usarkar)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] RTP payload list in OFFER
Hi Uttam,
Here is my SBC, even is not forwarding the call to leg 2... Its first
sending the 100 Trying in the correspondence of Initial Invite and then
immediately sendin
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Nitin Kapoor
Sent: Friday, July 30, 2010 1:36 PM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] RTP payload list in OFFER
D
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Vivek Singla
Sent: Friday, July 30, 2010 12:11 PM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] Request URI, FROM ,TO heade
Simon,
Why re-INVITE is having CSeq: 104 instead of 103 ( initial INVITE had
CSeq of 102)? This may not be an issue. Just wandering as you keep the
Call Id and other parameter same.
Why BYE has same tags in To and From header? If you fix the From tag
then it might resolve your issue.
f: ;tag=8
I don't think that's a good idea. First of all you are adding a new Header and
your falling in trap of an attacker.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
radhakrishna
Sent: Wednesday,
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