HI fellows,
I hope everyone be ok.
I would like someone can help me with the following.
I have been through the web trying looking for the RFC who talks about the
parameter values of the Authorization header in the SIP Register.
I have the following from one guide that I have, but I would like
Thanks a lot.
Do you have any idea what means the value 8084818088?
Regards,
Guillermo
De: Ranjit Avasarala [mailto:ranjitka...@gmail.com]
Enviado el: Friday, October 11, 2019 12:42 PM
Para: Zuñiga, Guillermo; Sip-implementors@lists.cs.columbia.edu
Asunto: Re: [Sip-implementors] PSTN parameters
specific PSTN parameters in a
> pstn-params= entry in outgoing INVITE messages. The PSTN parameters
> currently included in pstn-params are Screening Indicator, ISDN Indicator,
> and Transmission Medium Requirement (TMR).
BR
Philipp
Am Do., 10. Okt. 2019 um 22:45 Uhr schrieb
Hi Fellows,
I would like you can help me with the following:
I was trying to look for information about what means the numbers that appear
in some cases in the FROM header of the INVITE with the following
(pstn-params=8084818088).
From: ;tag=SDh25j501-gK0a3ed144
Could you help to identify whe
Hi Fellows,
I would like you can help me to be more clear with the following.
Why a INVITE arrive to a Class4 Control Switch could not be able to route a
INVITE that ingress without a 100rel supported parameter?
Example:
This is INVITE is not being routed by the C4.
Supported:
X-cisco-srtp-fal
Hi fellows,
I would like you can help me with the following doubt.
I am having a user is Registering ok to a Registrar Server.
Seeing the authorization values I can see that we have the Authorization Header
wiht the following Authentication-URI value= sip:domain.com
Realm = domain.com
But I am
Hi Fellows,
I would like you can help me to know if exist any restriction if in a INVITE
Request the Endpoint send me 2 mline with the same port, when I am using
TLS/SRTP.
Endpoint is sending the INVITE like this:
m=audio 40084 RTP/AVP 120 9 0 8 18 101
.
m=audio 40084 RTP/SAVP 120 9 0 8 18 10
Thanks for that,
I Access enviroment not PEER to PEER connection, CONTACT could be anonymous or
always should be any valid uri-user?
Regards,
Thanks
Guillermo
De: Aman [mailto:amanpreeet.si...@gmail.com]
Enviado el: Monday, February 26, 2018 3:11 AM
Para: Zuñiga, Guillermo
CC: discuss
Hi fellows,
I would you can help me to know and be clear about the following questions:
Is the CONTACT header mandatory that always have to be present in the INVITE?
I am asking that for the case that imagine that Endpoint after is Registered in
a Registrar Server, send me the following header i
Hi Fellows,
I am having the following issue with one device that is having one way audio.
The Calls is from a SIP UA to PSTN site, when the call is stablishing the only
way I have audio in both ways is when in the following case:
UASBC/GW->PSTN
INVITE---
mail: guillermo.zun...@cwpanama.com
From: Dale R. Worley [wor...@ariadne.com]
Sent: Tuesday, July 19, 2016 9:19 PM
To: Zuñiga, Guillermo
Cc: discuss...@sipforum.org; sip-implementors@lists.cs.columbia.edu
Subject: Re: [SIPForum-discussion] SIP to SS7 Calling Nu
Hi fellows,
Sorry Roman
Somebody know what means the following messages--->Message body is too big:
232253 bytes with a limit of 150 KB
Your mail to 'discussion' with the subject
RE: SIP to SS7 Calling Number Information
Is being held until the list moderator can review it for approval
Cel:+507 6670-0481
Email: guillermo.zun...@cwpanama.com
-Mensaje original-
De: discussion-boun...@sipforum.org [mailto:discussion-boun...@sipforum.org]
Enviado el: Tuesday, June 28, 2016 9:45 AM
Para: Zuñiga, Guillermo
Asunto: [QUAR] Your message to discussion awaits moderator appr
Hi Lennox I am trying with just this email and even like this I can get any
feedback, and I can´t see the info in the list.
Thanks for any feedback.
Hi fellows,
I am having the following issue with a peer carriers conected to my GW.
The scenario is the following, a call with a Invite which head
Hi Mrs
How can give a tips?
I am sending Calling from a SipEndpoint to a Fax at the PSTN,
And I am getting a SIP Unsupported media type messages and then the connection
is drop.
In the invite coming from a AplicationServer-AS to a VerazSoftswitch I am
seeing the parameter (a=silenceSupp:off -
lermo
From: Zuñiga, Guillermo
Sent: Friday, April 18, 2008 8:30 AM
To: 'Halit Sakca'; =?windows-1254?Q?Zu=F1iga; [EMAIL PROTECTED];
sip-implementors@lists.cs.columbia.edu
Subject: RE: [SIPForum-discussion] Differences between 2 Invite
SIP endpoint Veraz switch
21
8, 2008 5:51 AM
To: =?windows-1254?Q?Zu=F1iga; Zuñiga, Guillermo; [EMAIL PROTECTED];
sip-implementors@lists.cs.columbia.edu
Subject: RE: [SIPForum-discussion] Differences between 2 Invite
hey mate,
to investigate more efficient could you please let me be clear about the case?
"I am sendi
Mr,
Who could help me.
I am sending a sip call to the same mgw coming from differences IP-Carriers,
The first call with file "invite ok ethereal.txt" is a good call
And the other call with file "invite bad ethereal.txt" is a bad call.
In the good call I am seeing the parameter (Med
Hi Doctores.
I have 2 SBC connected, and I want to delay the timeout of the Call,
Cause the Far End is trying to ReRoute the Call until It got free circuits, I
am sending Cancel to this Call before he Completed it.
I was reading the RFC 3261 and I would like to match my Timers Value in t
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