Antonio Gómez Soto wrote:
snip
Ok, thank you. One final question:
I see that it's possible to have multiple auth's in an endpoint. For
incoming traffic to be
authenticated, how does pjsip know which auth to consider? By looking at
the From: address
in the SIP header, and matching that up
Hello,
I am slightly confused by the difference between chan_sip and pjsip.
Especially the new (to me) objects aor and contact.
I am having trouble mapping them to the typical SIP configuration settings
on a phone.
Suppose I have a phone with two line buttons, for two extension numbers.
Now,
I
Antonio Gómez Soto wrote:
Hello,
Kia ora,
I am slightly confused by the difference between chan_sip and pjsip.
Especially the new (to me) objects aor and contact.
I am having trouble mapping them to the typical SIP configuration
settings on a phone.
Have you looked on the wiki[1]? There's
On Sun, Jan 4, 2015 at 3:29 PM, Antonio Gómez Soto
antonio.gomez.s...@gmail.com wrote:
Hello,
I am slightly confused by the difference between chan_sip and pjsip.
Especially the new (to me) objects aor and contact.
I am having trouble mapping them to the typical SIP configuration settings
Thanks for responding,
On Sun, Jan 4, 2015 at 5:45 PM, George Joseph george.jos...@fairview5.com
wrote:
On Sun, Jan 4, 2015 at 3:29 PM, Antonio Gómez Soto
antonio.gomez.s...@gmail.com wrote:
Hello,
I am slightly confused by the difference between chan_sip and pjsip.
Especially the new
Antonio Gómez Soto wrote:
So basically, the 'contact' in the AOR is just an ip address (or
'dynamic', in which case it accepts
incoming registrations).
A contact is a SIP term, it's a way of getting to something. (IP
address+port)
So what happens if one endpoint has multiple AOR's which
Joshua,
On Sun, Jan 4, 2015 at 6:39 PM, Joshua Colp jc...@digium.com wrote:
[..snip..]
Also I notice, an AOR does seem do be directly correlated with an auth
record, so why are
they separate in the configuration, why not unify the aor and the auth
objects?
They aren't at all. Auth =
Antonio Gómez Soto wrote:
snip
I did not mean they are the same, I meant that there seems to be a
one-to-one relationship.
So I am wondering, since the auth does seem useless without an aor, but
an aor
can exist without an auth, why was the auth object created in the first
place,
instead of
On Sun, Jan 4, 2015 at 8:48 PM, Joshua Colp jc...@digium.com wrote:
Antonio Gómez Soto wrote:
snip
I did not mean they are the same, I meant that there seems to be a
one-to-one relationship.
So I am wondering, since the auth does seem useless without an aor, but
an aor
can exist
Hello list,
I read this in sip.conf :
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
What does this mean ?!
Does this mean that when I mark this as yes, a phone that already has
taken a call will be
Hi!
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
Does this mean that when I mark this as yes, a phone that already has
taken a call will be send a second and third call ?!
No, not directly: This setting is only
Hi!
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
Does this mean that when I mark this as yes, a phone that already has
taken a call will be send a second and third call ?!
No, not directly: This
I'm confused about something
In Asterisk 1.4 you can collect RTP QoS metrics at the end of a call with:
${CHANNEL(rtpqos,audio,all)}
Now, when your using the AMI to do a callout, like this...
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: SIP/1000
Variable:
I'm confused about
SIP realtime updates. If I make a database change, and then do a "sip prune
realtime peer peer", I can see Asterisk query the database, and retrieve
the updated information. However, it still uses the old values. What's up with
that?
If I do a "reload",
Your restriction is what the service provider allows. Most (that I've
used)
allow g729. I know it uses more bandwidth than g723 but nothing like G711
(ulaw or alaw) and from my experience, the quality is quite reasonable.
uh...g729 uses the least bandwidth
-jon
I though g723 used 5.43 KBps and g729 was 5.91 KBps
On 5/16/06, Jon Weisman [EMAIL PROTECTED] wrote:
Your restriction is what the service provider allows. Most (that I've
used)
allow g729. I know it uses more bandwidth than g723 but nothing like G711
(ulaw or alaw) and from my experience,
, 2006 23:36
Subject: Re: [Asterisk-Users] Confused !
Install iptraf, that will allow you to check incoming and outgoing traffic
(or trafshow what do that on /host basis, but not so detailed info)
If you choose ulaw, that should take about 90kbps fullduplex traffic.
I'd like to share
@lists.digium.com
Sent: Sunday, May 14, 2006 11:27
Subject: Re: [Asterisk-Users] Confused !
thanks for your replay,
after i disallow all codec except g723 i also confused how a2billing
is working then what i did , i removed all the codec from
/usr/lib/astersik/module without codec_g723.so .
then i
Salaque [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, May 14, 2006 11:27
Subject: Re: [Asterisk-Users] Confused !
thanks for your replay,
after i disallow all codec except g723 i also confused how a2billing
is working
Subject: Re: [Asterisk-Users] Confused !
how to use reinvite in my asterisk setup ?
thanks
Salaque
On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote:
I'm not an authority
but why don't you get some g729 codecs (10 or so) and use g729 all around.
Not allowing for ADSL overheads you can calculate your
Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, May 14, 2006 14:16
Subject: Re: [Asterisk-Users] Confused !
how to use reinvite in my asterisk setup ?
thanks
Salaque
On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote:
I'm not an authority
but why don't you get
Hello list,
I'd like to share something u all , so that i could understand whats
going on into my Asterisk box.
i have a setup like this
client(ip phone) -ip network--- [Asterisk]ip network
---[Service provider]
i have configured A2biling in my Asterisk box. so when client
Install iptraf, that will allow you to check incoming and outgoing traffic
(or trafshow what do that on /host basis, but not so detailed info)
If you choose ulaw, that should take about 90kbps fullduplex traffic.
I'd like to share something u all , so that i could understand whats
going on
the A2billing IVR working ? I have to assume
G711 (ulaw or alaw) is used.
- Original Message -
From: Woodoo People .pGa! [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, May 13, 2006 23:36
Subject: Re: [Asterisk-Users
AR Tarzi wrote:
Unless reinviting works, wouldn't that add up to what he's experiencing ?
client - asterisk - service provider.. makes that 180k each connection
so 4 of them would give 800k or so.
What I can't understand is: if only g723 is allowed, and Asterisk only
allows it as passthrough,
On Friday 21 April 2006 16:29, T. Shaw wrote:
After getting an account and following their steps, I can make calls out
using my IAX (cubix) and Sip (Xlite) phones.
However, I'm a bit confused on the purpose on how my box asterisk box is
involved. I completely turned off my Asterisk box, and
Hey guys,
I'm actively trying to get the big picture on how all this works and
relates to each other.
I've gone through some basic examples from the book and from the sample
files just fine.
Now, I've setup an account with a VOIP provider which does IAX termination
(exgn.net)
After getting an
The VOIP provider does not actually care if you are making your calls
from a simple SoftPhone, or a complete PBX. Im not going to explain
all the possible combinations of connecting, but i guess your
confusion comes because you still dont get to the part where some one
explains to you what Native
Very good.. I modified your things (although not much) so that if the
agent is logged in it automatically logs them out.. if they aren't
logged in then it prompts them to login.
On 4/4/06, Alan Ferrency [EMAIL PROTECTED] wrote:
The portion I forgot to mention:
Our agent login extension checks
The portion I forgot to mention:
Our agent login extension checks AGENTBYCALLERID to make sure no one is
already logged into the phone, before doing agentcallbacklogin. If you
don't do this, then it's entirely possible for two agents to be logged
into the same phone. However, only one will be
On Sat, 1 Apr 2006, Matt wrote:
However, anyone have a good way to log the agent out without having
them enter their agent ID and then have to hit # for the new
extension?
There are a couple of ways listed here in the Wiki:
Our solution to the agent log out problem is admittedly imperfect in
the general case, but it works well enough if you can do without agent
passwords.
- This only works if agents don't have passwords. You could probably
modify it to look up passwords in real time, but it wasn't important
However, anyone have a good way to log the agent out without having
them enter their agent ID and then have to hit # for the new
extension?
There are a couple of ways listed here in the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin
Hi,
I'm confused with agents and queues in Asterisk. If I use
AddQueueMember() then show queues shows the agents that I have
logged into the queue... however the agent ID has to be the extension
the agent is sitting at ... kinda useless for stats tracking.
If I use AgentCallbackLogin() then show
Hi Matt,
We have somewhat of a similar setup here in my office. We have multiple
queues to which different agents are a member to anyone of them.
Basically what I chose to do was make my own custom log in script. I
reference to the voicemail box and use the ID and password to
authenticate
Very good that's actually what I ended up doing.
I think my confusion came up with Hot Desk use ... like where you
want the agents EXTENSION to roam with them.. and agent-IDs.. which is
what I wanted... once I figured out the difference things have been
going smoothly.. While I'm a long time
On 3/31/06, Matt [EMAIL PROTECTED] wrote:
Very good that's actually what I ended up doing.
I think my confusion came up with Hot Desk use ... like where you
want the agents EXTENSION to roam with them.. and agent-IDs.. which is
what I wanted... once I figured out the difference things have
you can find some resources at the end of this page:
http://www.voip-info.org/tiki-index.php?page=ISDN
hope it helps.
best regards
On 8/14/05, Vedran Dakic [EMAIL PROTECTED] wrote:
Hello there.
What I'm trying to make is - have an asterisk server with
sccp/mgcp/skip/h.323
Hello there.
What I'm trying to make is - have an asterisk server with
sccp/mgcp/skip/h.323 support to handle calls
between various company locations. Let's say the company has
five different locations, internet connections
in each one of them and would like to use it via asterisk to
: [Asterisk-Users] Confused on G723 and G729
My question is.. if my voip terminator supports G723 and G729
only, do I still need a license?
Or is that considered
pass-through? If so, do I need to do anything special to
get it to work?
It is pass-through if both end points are using G
For instance.. when I try to use G723.1 on my phone (and just call in
from my PRI line) I get:
Unable to find a path from g723 to ulaw.
Unable to find a path from ulaw to g723.
No path to translate from Zap/1-1(68) to Sip/201-80c7(1).
Same things happens if I call in on my current provider's
In your case, where you will need the license is on the box that your
phones register to. For exampe, when someone checks voicemail,
encoding takes place, therefore you need a license.
Look at it this way:
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
- no license
So
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
This is how I'd be setup.. actually more like this:
[g729 provider] --(sip) [g729 asterisk
server](sip)---[g711 sip phone client].
So... if I understand this correctly.. I *would not* for *any* reason
On Thu, 2005-04-28 at 11:03 -0400, Matt wrote:
So
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
This is how I'd be setup.. actually more like this:
[g729 provider] --(sip) [g729 asterisk
server](sip)---[g711 sip phone client].
So... if I
I'll gladly pay $10 a license... I'm all for supporting digium...
however, I was under the impression that there was also some huge one
time fee of like $2,000 or something. I guess I was wrong... ok now
bad..
So I purchase the license from digium... then what happens/what needs
to be done on
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, April 28, 2005 8:31 AM
To: Adam Goryachev
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Confused on G723 and G729
I'll gladly pay
You would need a transcoding license between the Asterisk PBX and the
G711 phone...
On 4/28/05, Matt [EMAIL PROTECTED] wrote:
So
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
This is how I'd be setup.. actually more like this:
[g729 provider] --(sip)
I see that G723 and G729 require a license to be used, or can be used
(in the case of G723) in pass-through mode only.
My question is.. if my voip terminator supports G723 and G729 only, do
I still need a license? Or is that considered pass-through? If so,
do I need to do anything special to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, April 27, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Confused on G723 and G729
My question is.. if my voip terminator
Hi,
I'm confused whether I setup PC for Asterisk right or not. Module qozap is
alone (yet not sharing) in interrupt 209 (isn't this too high for native
interrupt).
Is this good state or not? If not, how to setup better ?
Thanks in advance,
regards,
Rob.
voip:~ # cat /proc/interrupts
I think I have got * worked out as far as getting users on a small
private network talking with each other, but when it comes to the bigger
picture about talking between private networks connected by the Internet
then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc.
Before I start
Are you going to use single E1 line? How many concurrent calls.
Single Xeon 3.0 with *1 GB *ram should serve 30-32 calls with not
problem, even during G729 transcoding and echo cancellation.
If you are not plaining to do G729 transcoding - then I believe you can
put more then 32 calls in this
Having kept an Eye on hardware specs mails in the list, and having read the
hardware notes on voip-info, I am still confused over the recommended
hardware:
We are going to have up to 100 operators and staff making calls, mostly
through zap connected to TE405P through to a 32-channel E1. However,
Hi all,
I´m using X-Lite as
SoftPhone in Asterisk. I have configured this:
[101]
;Turn off silence
suppression in X-Lite (Transmit Silence=YES)!
;Note that Xlite sends
NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=jozeph
callerid=Jozeph
Brasil 5678
Hi,
I have two * box. One is box1 and the other is box2. And I have two iax
clients A and B. A is registed with box1 and B is registed with box2. If I
make a call from A to B using the following method:
IAX/[user[:secret[EMAIL
PROTECTED]peer[:portno][/exten[@context][/options]]
The
-Users] Confused with CallerID when using the iax
chanenls
Hi,
I have two * box. One is box1 and the other is box2. And I have two
iax
clients A and B. A is registed with box1 and B is registed with box2. If
I
make a call from A to B using the following method:
IAX/[user[:secret[EMAIL
Hi!
I'm confused as to where I need to place the Asterisk server with repect
to my Linux Firewall? I've read thru the message archives but have not
been able to glean a clear answer. My Linux Firewall is a RH9 running
IPTABLES doing NAT.
The probably easiest solution is to put * onto your
Philipp von Klitzing wrote:
Hi!
I'm confused as to where I need to place the Asterisk server with repect
to my Linux Firewall? I've read thru the message archives but have not
been able to glean a clear answer. My Linux Firewall is a RH9 running
IPTABLES doing NAT.
The probably easiest
Title: Confused about Asterisk server with regards to Linux NAT Firewall
Hi all,
I'm confused as to where I need to place the Asterisk server with repect to my Linux Firewall? I've read thru the message archives but have not been able to glean a clear answer. My Linux Firewall is a RH9
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