Michael Devenijn wrote:
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ...
sip.conf extract :
[gw001]
type=friend
host=dynamic
defaultip=192.168.0.12
nat=no
dtmfmode=rfc2833
canreinvite=yes
qualify=no
context=tlsgw
I have a network of IAX servers connecting to each other. I just realized that IAX
does some
clever magic by itself. Let me explain:
---
Let's say you have three servers: A, B and Q
A calls B with IAX2
B connects the call to Q with IAX2
B realizes that
Barry Fawthrop wrote:
From: Olle E. Johansson [EMAIL PROTECTED]
snip
Check the CDRuserfield - it's a free field in the CDR you set in the
dialplan or from a script.
How would you set the CDRuserfield from the dialplan
exten = ?
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd
Barry Fawthrop wrote:
For some reason MWI, wants to dial [EMAIL PROTECTED], I have not exten
or
account asterisk ???, can't even find where this is set ?
http://www.voip-info.org/wiki-Asterisk+phone+snom
/O
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Matteo Rancilio wrote:
I'm using asterisk and it works ok the only thing is thata ecvery 2/3
days get the cpu load up to 99% and the only way I can shutdown the
service is to use a killall -9 asterisk.
any suggestions?
Nope, but you have to provide us with more information about your
http://bugs.digium.com/bug_view_page.php?bug_id=899
A patch that improves the DTMF support for ISDN4Linux and adds functionality for
CallerID handling with EuroISDN networks.
This patch needs testing and comments on the bug tracker. If you're using ISDN4Linux
(not CAPI) and have spare time to
I'm trying to set up my Eicon Diva ISDN card for outbound calls on my *.
Could someone mail me example configuration files off list, since there aren't
many examples in the manual.
Thank you!
/O
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Vonage got Cisco to include a password protect the config in the
latest version of the firmware, and as far as I know now all the Vonage
ATAs are forever destined to be used with Vonage and only Vonage.
Cell providers do the same, but they help you unlock the phone after
a set period - one or
XISCOAIR wrote:
Hi everybody,
I'm trying to develop a web application for controlling if SIP users
are registered in * or not, and show it in a web.
My problem is that I don't now if it's possible to do a Shell Script to
control this:
1. Connect to console.
2. Execute command.
3. Obtain users
Bruce Marler wrote:
All,
I have been beating my head against a wall trying to figure out how I would
implement a separate moderator code and participant code for the same
conference using meetme, the deal is I dont want the participants to be able
to join until the moderator is in the conference.
sip://[EMAIL PROTECTED] works perfectly well...
There are many benefits in stability when you use SRV records to find a
SIP proxy. However this requires that you have some sort of load balancing
between the servers.
It was a long time ago anyone mailed [EMAIL PROTECTED] Let's hope
we can prove
Duane wrote:
Olle E. Johansson wrote:
If you do not enable SRV records, you can't phone me. There's no
SIP proxy on edvina.net ;-)
Exactly my point, by ***DEFAULT*** Asterisk won't use SRV records, even
if it did, it doesn't support SRV correctly (as you pointed out), and
using an A record
This week, I've been really busy with the launch of a new Swedish Voip provider,
www.bbtele.se, so I haven't been able to follow the Asterisk community and haven't
been very responsive either. My apologies if you've tried to contact me and I did
not reply quickly or at all.
So to cover up (can't
Duane wrote:
If they want a simple method
of allowing calls they should use enum, least then it's obvious that it
isn't a email address and that they would possibly need to enable a few
things to make it work.
Enum doesn't replace SRV records at all.
Enum records point to a SIP URI.
To resolve
Time for Duane to start implementing DNS SRV, since it's from now on is turned on
by default in CVS head.
Thank you, Mark!
/O ;-)
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Rich,
I'm working on the tutorial agenda today. I've schedule a tutorial for you. Will need
a photograph
as well as some text for the web page that describes our tutorials.
Please read
http://www.astricon.net/astricon2004/tutorials.shtml
And you'll see what I need from you.
If you have any
List,
Sorry for sending a private e-mail to the list. Tired...
While speaking about Astricon, we are looking to fill the last holes in the
tutorial agenda.
We agreed on two topics that we feel are missing:
* Dialplan tips and tricks
* Agent and call queues
If you are interested in teaching one of
Senad Jordanovic wrote:
brian wrote:
That's the only way to make it work.
Devices behind nat, on same network, can call each other ONLY if
canreinvite is set to no? Is that what you are saying?
Canreinvite=yes *only* works if all devices are on the same side of the NAT, the
outside or the inside.
Doug Kennedy wrote:
Hello,
I have modified the VoiceMailMain application to satisfy the request of
the local users, i.e., my wife. She lost patience with too many
options (we have one mailbox, so we don't need to forward messages, or
reply to messages, or file them in 6 different folders...)
Thank you very much for all feedback on Asterisk Sunday News!
This is the last issue for June. This week I'll go on holiday
and will be back with more news in early July.
My kids are getting summer leave this week and we'll be
visiting the south of England for a while. Another part of
Europe that
Nicholas Bachmann wrote:
Olle E. Johansson wrote:
The decision is to base the future 1.0-release on the CVS head tree.
The current stable-1.0 tree will be released as something intermediary,
maybe 0.91, and at that point it will be considered end-of-life.
At some point when we have cleared the bug
Due to the dismissal of the stable-1.0 cvs source code, I've changed policy of the
Asterisk
Wiki - we now document CVS head. I would like all contributors to document which
version of
Asterisk (date) an addition was applied to, so readers can find out if a new function
works
with their version
Steven Critchfield wrote:
On Mon, 2004-06-14 at 11:13, Peter Mitchell wrote:
I can't seem to find the link to examples of asterisk installations for
different sized sites. I'm not after specific configuration of the conf
files, just an overview on what hardware/chassis cards people are
running
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing
Eric Wieling wrote:
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
At that pint ${EXTEN} is i. Try using ${INVALID_EXTEN}
Eric,
Thank you, I've added that
Andres wrote:
Ernest W. Lessenger wrote:
Can anyone tell me how (and for how long) asterisk remembers the IP
address
for an IAX2 peer? Voicepulse has been going up and down for me, and it
seems
to have something to do with the IP address changing. Is there a way to
force asterisk to run
chouck wrote:
Im having a trouble understanding the config files setup even with some
documentation ive read such as the handbook, maybe im just illiterate.
Anyway do you think some one can point me to some examples of real
config files. Such as IAX, Extensions, and Sip. I just cant grasp
Lenny Tropiano / asterisk.org Mailing list wrote:
We're doing some SIP development and have a question on additional parameters
supplied to the register (in this case maddr= and the non-standard clport= in
our example below).
What we're experiencing is the INVITE doesn't included these parameters
Sunday news is today published on a monday. Yesterday was fourth of
july, and I used that as an excuse for being off line yesterday.
(Sweden's national day is June 6th - and it's not yet a public holiday,
btw). Most of my Asterisk time lately have been used for producing
the registration site for
://lists.digium.com/mailman/listinfo/asterisk-users
--
Olle E. Johansson, Edvina.net AB, [EMAIL PROTECTED]
- Phone +46 8 594 788 10, Cell phone: +46 70 593 68 51
- IP phone: sip:[EMAIL PROTECTED]
- Address: Runbovägen 10, SE-192 48 Sollentuna, Sweden
- Web: http://edvina.net
Andrew Thompson wrote:
Peer: A connection that sends calls to asterisk.
User: A connection that asterisk sends calls out to.
Friend: an attempt at a combination of both, to simplify set up of phones
that send and receive calls. (There are several people here who will tell
you friend is evil.)
3) Can anyone explain the meaning of peer, friend,
user in more details? For each case, what is the
role of Asterisk in SIP world, a UA, a proxy, or
others?
In some diagrams, Asterisk take's the role of a SIP Proxy, but it is *not*
a SIP proxy by design. Asterisk answers SIP calls and originates
Registration to Astricon - the first Asterisk user's and developer's conference -
is now open. Astricon is taking place at the Atlanta Marriot September 22-24.
Digium is our Diamond partner in arranging this conference.
The web site is updated with information on hotel, prices and speakers for
the
Paul Mahler wrote:
Well, this is certainly getting exciting.
Yes, it is. Sorry for coming in late to this debate...
Andy, I took your advice and re-read the RFP.
It's actually RFC, not RFP. (teasing :-)
So, gentlemen, help me out here. The spec says:
The Address of record is the SIP address
Kannaiyan Natesan wrote:
* No, there's no quick fix for a 100 USD bounty
How much you estimate on quick fix?
I apologize for my Swenglish language...
I don't believe there's a quick fix at all.
If you want a quote for a fix, contact me off-list. But remember, that I believe
that fixing this is
You have not shown us ANY example yet for which this
facility is *NEEDED*.
Well, I have users that get an account on my PBX.
I really don't care how many phones they want to use, hardware phones on
their desktop or soft phones on their laptop while travelling. It's still a user
with one account.
Sunrise Ltd wrote:
Olle E. Johansson wrote:
Well, I have users that get an account on my PBX.
I really don't care how many phones they want to use,
hardware phones on their desktop or soft phones on their
laptop while travelling. It's still a user with one
account.
Two words: self provisioning
Nathan Alpert wrote:
Sorry to ask this question here since it's related to IRC and not
Asterisk, but I am having trouble logging into the #asterisk IRC channel
on freenode and was wondering if anyone else has had this problem and
solved it.
So here's the situation: Whenever I try to login to the
Holger Schurig wrote:
I keep replying to myself quite often.
As it turned out, this is a problem with incrementing CSEQ on the
Grandstream. I don't have the clue if the SIP specification says that you
have to increment it, but the GS sometimes sends a different SIP message
with the same CSEQ.
http://bugs.digium.com/bug_view_page.php?bug_id=0001858
Constfilin writes:
The attached patch allows dynamic configuration of asterisk queues.
Queue information is re-read from the configurable database in real time.
Additional Information Right now implemented only for postgres, no mysql
Please
http://bugs.digium.com/bug_view_page.php?bug_id=0002013
If you use the Czech language, please test this and add your
opinion, good or bad, to the bug tracker.
/O
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http://bugs.digium.com/bug_view_page.php?bug_id=0002055
This patch adds the ability to send text and HTML messages as
voicemail notificiations.
Please test and respond to the bug tracker!
/O
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http://bugs.digium.com/bug_view_page.php?bug_id=0001693
This patch adds a lot of options for AgentLogin/AgentCallbackLogin
Please test and respond in the bug tracker!
/O
-
This patch adds quite a few new features
http://bugs.digium.com/bug_view_page.php?bug_id=0001188
This patch unifies the code that decides on the location of a mailbox and
stores voicemail in a tree-like structure, to be prepared for very large
volumes of voicemailboxes in one file system.
It's disputed whether this affects performance on
Nicolas Gudino wrote:
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
chan_sip2
This week starts with the exciting news: We're getting close to
Asterisk 1.0 again. After the failed attempt earlier this year,
we've been able to remove a lot of the MAJOR/CRASH bugs from the
bug tracker and Mark feel's it's time to target 1.0 again.
At this point, the community needs to work as
muralikrishnan lakshmanan wrote:
Hello friends,
I got one page from net http://www.voip-info.org/wiki-CLASS;
In that page I saw lot of *xx codes for asterisk feautres.
I don't know how to use these codes.
If anyone used these codes can you teach me.
This is just a list of
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing
Matthias Endler wrote:
As promised yesterday:
Anybody interrested can download the patch for Asterisk 0.9.1 at
http://matthiasendler.net/asterisk/patch/.
Great!
Please add it to the bugtracker in a .txt file created with
cvs diff -u channels/chan_sip.c
The diff has to be for CVS HEAD, that is
-submit it to bugs.digium.com.
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Holger Schurig wrote:
I think we have several problems here. Once it's Peer:, the other time
it's Peername.
That's clearly a bug.
Also, I don't like the name of the event. It should just
be PeerStatus and PeerRegistration, because we might add something to
IAX2 as well. So I'd suggest to do
Holger Schurig wrote:
And while I was at this patch, I also changed the
Event: SIPRegistry
Domain: ...
Status: ...
to
Event: Register
Channel: SIP
Domain: ...
Status: ...
I still believe it would be better to call this Registry since that's a common
term across IAX and SIP for outbound
John Todd wrote:
I hate being a me too poster, but the double-hash patch I have
implemented four times now, and I know at least three other people who
have also gone well out of their way to put that patch into their
system. Making this an official modification would be ideal, in my
opinion,
But, you could use a third-party fax thingamajig and I'm sure connect it
to * for a good UM solution.
Just pass it to hylafax and you fly, but it requires some planning cause you
will need ddouble the amount of ports plus the fax devices for hylafax
Interesting - please, do you have time to
Tomas Prybil wrote:
max power wrote:
Spent that last week or so trying to get isdn4linux working. how
do I link ttyIO to asterisk?I cannot dial out or dialin. I can
see the call coming in in /var/log/messages.
Has anyone any tips? I am not familar with isdn4linux.
What kind of
Hmm. this rings a bell, try putting nat=yes in your sip.conf, I think that fixed
the problem for me. (Or was the the login timed out thing? *shrug*)
The manual is not very clear on what happens with nat=yes in sip.conf.
Anyone here that could write a simple explanation of this option?
/O
All of us SIP users have problems with NAT boxes from hell... :-)
There's some components that needs to be documented, like
* What's STUNs role and how do we implement it alongside with NAT
(maybe Vovida.org stun server)
* What is the function of NAT=yes in sip.conf ;-)
* Do I have any use of
Tomas Prybil wrote:
Olle E. Johansson wrote:
snip
Everyone points to capi and, back to the start of my reply, it seems
expensive for personal
use...
A passive AVM Fritz card is somewhere around 100
I rest my case, sir :-)
Only looked at Eicon and other cards available up here in the cold north
Rich Adamson wrote:
Could someone give me a 10,000 foot view of what the differences are
between Ser and Asterisk?
Asterisk is a PBX that you can use to connect SIP clients to the PSTN
or voicemail /IVR applications.
SER is a SIP proxy that connects SIP clients to each other.
Asterisk handles all
I've spent some time on the Wiki adding documentation on all asterisk applications from
the cli 'show application ' commands. I've also added some cross references and
pointers.
http://www.voip-info.org/tiki-index.php?page=Asterisk
If you find this useful, please go there and help us build a
Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.
I
Steven Critchfield wrote:
I've added a security page to the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+security
Maybe there should also be a link for best practices with respect to
dial plan layout.
I guess since this is my second comment on the wiki, I should log in and
Tilghman Lesher wrote:
On Wednesday 10 September 2003 10:51 am, Olle E. Johansson wrote:
Lubomir Christov wrote:
today I found this security report regarding Asterisk SIP
Security.
http://www.securiteam.com/securitynews/5LP0720B5G.html
Important information. Why a silent patch
I took the liberty of adding Leif's FWD Asterisk configuration to the WIKI, so for an
- yet incomplete- overview on how to connect Asterisk and FWD please go to
http://tinyurl.com/mwe0
And, please add info or mail me configurations that works for connecting to FWD with
the asterisk server
A new RFC was published today, RFC 3601:
Abstract:
This memo describes the full set of notations needed to represent a
text string in a Dial Sequence. A Dial Sequence is normally composed
of Dual Tone Multi Frequency (DTMF) elements, plus separators and
additional actions (such as wait
John Todd wrote:
I'm using a Cisco 7960 with asterisk and any recording
on the machine, be it voicemail prompts, time of day,
echo test message, etc, is cut off for the first 1/4 to
1/2 second. I've tried setting the phone to gsm but
it still happens.
Before running any application that has
Try the CLI command:
SIP debug
...and you'll propably see that the FWD SIP server fwd.pulver.com answers something
like
ough... That's ugly...
It's referring to the fact that you're trying to communicate with a private IP
address (192.168.x.x).
With SIP clients like Xten, you configure
I've tried to find documentation on if Asterisk supports DNS SRV records for sip servers.
Reading the source of channel_sip.c it seems not:
hp = gethostbyname(hostname);
if (!hp) {
ast_log(LOG_WARNING, Host '%s' not found at line %d\n, hostname, lineno);
return -1;
}
Steven Critchfield wrote:
On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote:
While on the subject of Voicemail - what is the difference between
voicemail() and voicmail2() ?
From the application stand point there is little difference, but from
the configuration stand point there is a fair
SER, the SIP Express Router, version 0.8.11 have a new solution for supporting NAT,
the nat helper module.
See http://www.voip-info.org/tiki-index.php?page=SER+nat+support
It mangles the SIP header, the SDP and also actively keeps the NAT assosciation for
clients behind NAT open in order to be
Paul Cheng wrote:
On Friday, September 12, 2003, at 08:47 PM, Eric Wieling wrote:
Does anyone know how you specify MD5 auth on a register = line?
You have to specify it in sip.conf in the entry for the UA/proxy for
which you want to register with.
For example, if you want to register with
Zara Trousk wrote:
( Nobody explained me properly why the code was not developed but as you know Asterisk
is Digium and
Digium makes voice boards, so... In other words, what they are saying is: Buy Digium.
I think that's unfair. Asterisk is Open Source - everyone's free to add or change
stuff
PJ Welsh wrote:
I have to defend us newbies on this.
This environment does not facilitate sequential knowledge building!
You do realize that the http://www.asterisk.org/index.php?menu=support lists the mailing list first for support, don't you. In fact, you have to go to the second page before
Rémi Letot wrote:
Olle E. Johansson [EMAIL PROTECTED] writes:
couic
I realized the same and started a process to collect a lot of that
information and build a knowledge base on http://www.voip-forum.org/
Everyone is right, it should be http://www.voip-info.org
I confused with my attempt
Look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysql
/Olle
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James Sizemore wrote:
Has anyone browsed through the source code and
made a list of menu option for VoiceMailMain2?
Or know of some user documentation hiding
in Internet land some place? If not there well
be soon. Ho hum.
Start here
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain2
Also
Trying to read and understand bits and pieces of chan_sip.c I've found these I would like someone to clarify:
* srvlookup=yes|no
* pedantic
* canreinvite=update|yes --update seems new
Being curious, especially for srvlookup functionality...
/O
___
[EMAIL PROTECTED] wrote:
Michiel Betel [EMAIL PROTECTED] said:
Does there exist a text file with all the 'standard' Asterisk voice
messages? I'm planning to get them recorded in dutch, but need to know the
exact text of each prompt...
Michiel - are you planning to release the recordings? I had
Tais M. Hansen wrote:
On Wednesday 24 September 2003 16:36, Jamie Carl wrote:
Lack of documentation?
Welcome to the bleeding edge...
I know, I just meant that pretty much everything else is either descriptive or
described in sip.conf. Except the meaning of [xxx] entries.
Logged in this morning to find that the sound file scripts are now up on the Wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files
Courtesy of Zac Sprackett, I believe. Thank you!
/Olle
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I got curious of this function and tried to summarize by reading your mails and looking into the source code.
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
As you see, there are some commands available in the call file that I could not figure
out.
If you have figured these
Brancaleoni Matteo wrote:
VoIP protocols normally use 2 connection:
* 1 for control (eg on port 5060 for sip)
* 1 for the RTP (media stream)
The latter hasn't a fixed port, since is negotiated
by the control connection. That could cause some troubles
with NAT firewalls.
IAX doesn't use 2
Here's a good example: wiki? there is a wiki? :-) I saw a broken link
once but heard no more...does it still exist?
Oh yes, there is a Wiki.
And there's a lot of how to's and software documentation up there.
If you search the archives, I've mentioned the correct URL so often
that people propably
Take a look at the switch app..
Also search the archives for switch..
Thank you, I've missed that function.
Found an example in the archives and in the sample extensions.conf in the distribution.
Updated the wiki ;-)
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
WipeOut wrote:
Olle E. Johansson wrote:
I still can't get Windows messenger to register with a secret to
Asterisk.
Anthony - do you connect without registering or does Windows messenger
register properly with your * ?
/O
Have you tried forcing Asterisk to use plain text authentication
Armand A. Verstappen wrote:
I think we should have these setups listed:
- home user with 1-2 telco lines and 2-5 phones
- small office with 4-8 telco lines and 8-16 phones
- small office with a fractional E1/T1 and 12-24 phones
- medium office with full E1/T1 and 24-48 phones
- medium office with
Tilghman Lesher wrote:
On Monday 06 October 2003 05:13 pm, Carlton J. O'Riley wrote:
Are there any plans to incorporate the running of Asterisk as a
non-root user into the current CVS? There is nothing in Asterisk
that requires root access as far as I know and this would solve the
vmail.cgi
Leif Madsen wrote:
think of things I've missed, feel free to chime in!
Leif, why don't you put the script up on the wiki so that we all can
edit it and add on line with versioning?
As a newborn Asterisk user, I had severe problems configuring an ISDN card.
I believe a lot of new users start with a
WipeOut wrote:
duncan wrote:
actually i meant how to find out how many i could push down the 512k
line - with regards to codec bandwidth and signalling etc...
Measure the data rate on one call and divide 512k by it for a rough
estimate..
If you want more accuracy make one call and measure
Alastair Maw wrote:
On 13/10/03 14:05, Conrad Braun wrote:
Why do you want to remove some of the conf files? Just leave them
all there.. its not like they use up a lot of space or anything..
:)
I am just starting to use asterisk as well as VoIP in general, and
it's a bit confusing finding out
Chris Albertson wrote:
This is the big problem with using Asterisk for SIP. With Asterisk
the audio data between two SIP extensions has to actualy go into
then out of the Asterisk box. This does not scale well to
thousands of users like in a university campus or a comercial
SIP service.
My main question lies in the interworking between iptel's SER and Asteriks.
Not only on the configuration side, but also on the network side (here I
mean: can both run on the same server, or do they need to have different IP
addresses, ...).
My 10 cents:
Make sure that you run the two SIP
Anton Tinchev wrote:
Adam Hart wrote:
Sent: Wednesday, October 15, 2003 1:06 PM
Subject: [Asterisk-Users] Digium should develop and sell just Dummy card.
For timing...
Doesn't ztdummy already do this?
Only if you has the right usb chip
And the winner is? :-)
There's no comment in the source code
Asterisk will bridge a call in some cases and not in others. If codec
conversion is required between phones, its stays in the middle. If the
two phones can agree upon a common codec, etc, * is not in the middle
from a pure communications perpective. In that particular case, what
the phone does
WipeOut wrote:
One for the gurus..
Obviously not for me, but I'll dare to give it a shot anyway ;-)
Anyway, I decided to go and have a quick read through the SER docs and
in the section about NAT they say that the best way to address NAT is to
use STUN or uPNP..
STUN is helpful, but as I
WipeOut wrote:
Olle E. Johansson wrote:
WipeOut wrote:
One for the gurus..
Obviously not for me, but I'll dare to give it a shot anyway ;-)
Anyway, I decided to go and have a quick read through the SER docs
and in the section about NAT they say that the best way to address
NAT is to use
John Todd wrote:
Is there a way to tell Asterisk to use a SIP proxy?
For example I need everything going out to [EMAIL PROTECTED]
to be sedt to a proxy called proxy.foobar.com
I assume this should go into sip.conf but I don't see anything
in the documentation. Is this what the host= is for like
John Todd wrote:
Olle wrote:
STUN is helpful, but as I understand it analyzes the situation and
reports
the configuration of a NAT. It doesn't help you keeping the NAT
session open,
as SER module nathelper or the FWD/Jasomi solution.
Check here http://www.voip-info.org/wiki-SER+module+nathelper
I've added info on qualify=yes and how this can help the NAT dilemma to the
wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+qualify
As Wipeout stated, this is extremely useful and an important part of
the undocumented Asterisk features we're trying to document on the Wiki ;-)
I
WipeOut wrote:
duncan wrote:
I've added info on qualify=yes and how this can help the NAT dilemma
to the
wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+qualify
As Wipeout stated, this is extremely useful and an important part of
the undocumented Asterisk features we're trying
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