Hello,
Is it allowed by PBX on customer premises for a SIP trunk to change the port
number on VIA header once 401 UNAUTHORIZED received from SBC (via call
controller) for an outbound call ? The UDP port initially was 15275 which was
changed to 5060 after 401 response.
Before 401 (origina
Mahesh,
Thanks for the detailed response, very helpful ...
--- On Fri, 10/1/10, Maddipatla, Maheswara Rao (Mahesh)
wrote:
From: Maddipatla, Maheswara Rao (Mahesh)
Subject: RE: [Sip-implementors] Question about significance of
header(Supported: timer, 100rel)
To: "Rashid Shakil&qu
s has 100rel enabled. Thanks ...
Supported: timer,100rel
- Rashid Shakil
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Dale,
Thanks for the comments. I will check RFC 3263 for more details.
- RS
--- On Thu, 5/27/10, WORLEY, Dale R (Dale) wrote:
From: WORLEY, Dale R (Dale)
Subject: RE: [Sip-implementors] Port with SIP REQUEST-URI
To: "Rashid Shakil" ,
"sip-implementors@lists.cs.columbia.edu&qu
Greetings,
Quick question is transport port is a mandatory requirement with REQUEST URI
for INVITE and REGISTER requests ?...Are both of the below mentioned formats
are correct ? One is with transport port mentioned with RURI and another one is
blank ...
REGISTER sip:hrnd-sip.vssi.rashid.com:
Thanks for the reference Aneesh ..very helpful
--- On Thu, 2/25/10, Aneesh Naik wrote:
From: Aneesh Naik
Subject: Re: [Sip-implementors] SIP Media Flow Attribute question
To: "Rashid Shakil"
Cc: "Brett Tate" ,
"sip-implementors@lists.cs.columbia.edu"
Date: T
they send media stream without sharing SDP ? Later on
they send 18x with SDP and that's when media stream flowing in both direction.
- Rashid Shakil
--- On Thu, 2/25/10, Brett Tate wrote:
From: Brett Tate
Subject: RE: [Sip-implementors] SIP Media Flow Attribute question
To: "Ras
Hello,
Quick question ...If a SIP UA sends an INVITE without specifiying the media
flow direction attributes (sendrecv, recvonly, sendonly, and inactive). What is
the default behavior for the receiving gateway in setting up media flow ? Means
how does the far end gateway respond if they receive
Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Rashid
Shakil
Sent: Friday, December 11, 2009 12:20 AM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] Query about Media (RTP) ri
its documented anywhere ..Thanks in
advance for help.
- Rashid Shakil
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Walter,
I have seen this format of Diversion at my core. Not sure if quotes are allowed,
Diversion: ;reason=no-answer;privacy=off;counter=3
Diversion: ;reason=no-answer;counter=3;privacy=off
RS
--- On Tue, 8/11/09, ZEGELS Walter wrote:
From: ZEGELS Walter
Subject: [Sip-implementors] Ques
Hello,
Quick question please, following is an INVITE for a redirected call. Means
original call from PSTN gets redirect to another offnet number. If I am sending
this call to a SIP peer can you please tell me which headers (RURI & TO or TO &
FROM) are they going to follow to make call catego
icted
call (caller ID hide)
To: sip-implementors@lists.cs.columbia.edu
Date: Tuesday, December 16, 2008, 2:55 PM
El Martes, 16 de Diciembre de 2008, Rashid Shakil escribió:
> I am wondering for restricted calls carrier have to send "CALLING
FROM"
> information (like RPID with Priva
6/08, Iñaki Baz Castillo wrote:
From: Iñaki Baz Castillo
Subject: Re: [Sip-implementors] Remote Party ID (RPID) missing for Restricted
call (caller ID hide)
To: sip-implementors@lists.cs.columbia.edu
Date: Tuesday, December 16, 2008, 1:20 PM
El Martes, 16 de Diciembre de 2008, Rashid Shakil esc
ID hide)
To: sip-implementors@lists.cs.columbia.edu
Date: Tuesday, December 16, 2008, 12:43 PM
El Martes, 16 de Diciembre de 2008, Rashid Shakil escribió:
> Quick question please ...Can any SIP peer allowed to send an INVITE for
> restricted call without calling FROM information with no
ndrecv
a=maxptime:20
==
Regards,
Rashid Shakil
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Hello,
While working with SIP carrier I came across to an issue where particular SIP
providers rejecting my calls to a cell phone number. The call happen to be a
redirected call (means inbound call redirected to an offnet TN). For the
redirected call (call to an offnet number) our AS is send
Hello,
Folks just have a quick question. If I am receiving a caller ID
hide/anonymous call what should be privacy header (part of Remote-Party-ID
header) look like in the SIP peering environment. Should the privacy header is
"privacy=full". One of my SIP Peer is sending "privacy=uri" for a
map" is not mandatory for static
payload types. Payload type 0 belongs in this range and corresponds to
G.711 uLaw. Similar is the case for payload type 18 but not for 100
which is a dynamic payload type.
On Tue, May 20, 2008 at 8:51 PM, Rashid Shakil wrote:
>
> Quick question I was anal
Quick question I was analyzing the SDP which my softswtich is sending in
response to an INVITE for an inbound call I received from my SIP peer. I
noticed that I can see two codec choices in media description (m=) line but the
rtpmap attribute of the SDP just showing one. Can any one tell wha
we can redirect a call with diversion header and without SDP. Is
this explanation from my upstream provider is correct do I have to send an SDP
for the redirected call
Please reply me so that I can move forward with this
issue.
Regards,
Rashid Shakil.
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