Klaus Peras wrote:
Hey there,
does anybody know a CLI SIP Client für Linux?
I think you may find one in Vovida.org
/O
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[EMAIL PROTECTED] wrote:
hi,
Is asterisk a registrar server.
It all depends. If you mean registrar for Inter-Galaxy Travel
Permissions, no. If you mean SIP registrar, yes.
But we are not a SIP proxy ;-)
/O
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These OPTIONs packets are what we send for qualification, a scheme that
can also be used for NAT keepalives. You turn them on by adding
qualify=yes in the [peer] section of sip.conf
With qualification on, we regurlarly measure the latency between
Asterisk and the client and decide whether the
Rich Adamson wrote:
Cross posted on purpose
FYI, just upgraded from cvs-head from March 23 to this morning (March 31).
All compiles and installs completed normal.
Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the
standard oche... message. Piped the output to a text file
During the developer's conference call yesterday evening,
it was decided that we finally should release the much-awaited
Asterisk 2.0 Stable release, also called codename AAFJ.
This relaese is based on the hidden cvs that has been in
operation for six months by a group of core development members
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
Ronald Wiplinger wrote:
I would like to get a notice by email, if we run out of gateways!
exten = _9011Z.,410,Busy
exten = _9011Z.,411,EMAIL = How to?
-= Info about application 'System' =-
[Synopsis]:
Execute a system command
[Description]:
System(command): Executes a command
administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck
Brian West wrote:
OpenVPN
What happened to AES in IAX2?
/O
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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
Serge Schumacher wrote:
Can it be that the MeetMe application is not installed by default even
if there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)
It is not installed if you haven't got a Zaptel timer. See the Wiki docs
on
Mark just committed a small fix of mine to FastAGI. Previously there was
a script option to the URI that wasnt't used. Now, it's sent to the AGI
server so that one running server can handle multiple AGI functions.
agi://hostname:port/script
is the full syntax for the fastagi option to
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
Geoff Speicher wrote:
Sipura has implemented auto-answer in version 0.9.5 of the SPA-841
firmware. However, it is implemented via the Call-Info header, which
Asterisk stable doesn't currently support.
The attached patch implments a quick hack to support the Call-Info
header from the Dial()
Robert Spielmann wrote:
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for [EMAIL PROTECTED] the call is
Andrew Thompson wrote:
I am about to start a program that will be generaging sip device
configurations for sip.conf. My current sip.conf contains friend entries
for each SIP device connected to asterisk.
Should I even be attempting to split these in to seperate user/peer
devices?
Can two
Vlasis Hatzistavrou wrote:
Hello,
We hve been trying to make Asterisk work with SIP proxies with no success.
Is there support for SIP proxies in Asterisk in the latest versions?
A lot of people use Asterisk with SIP proxys.
What is your problem, give us a bit more information.
/Olle
I'm sitting in a hotel close to the Madrid airport... Any Asterisk users
in the neighbourhood that wants to meet me for a beer and some Asterisk
hacking this evening?
Send e-mail to me *off list*, thank you.
/O
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Asterisk wrote:
I've got a test * server (hppbx) where I install CVS-HEAD as often as
possible, with my extension registered to this, talking through IAX to
our production server which then channels out to the PSTN.
After completing a call just now, the following appeared on the CLI of
hppbx
Peter Svensson wrote:
On Wed, 16 Feb 2005, Rob Scott wrote:
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
Asterisk clocks outgoing rtp data to
Chris St Denis wrote:
I am using mysql sipfriends and can't seem to get the MWI to work. From what
I've read it seems this is not supported with that dynamic system, and
probably never will be.
In the 1.0 stable release, you can not send MWI for database peers.
In CVS head, the base for the future
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
Marcello Lupo wrote:
Hi,
notice that i have Grandstream phones and i have the problem if i activate the
Send Anonymous function on them.
If i do not activate that option the ACCOUNTCODE is correctly populated. SO i
think it may be a bug of asterisk.
I'm using Asterisk CVS-HEAD-10/07/04-18:07:25
Kevin P. Fleming wrote:
I have a patch in my local system that allows the canreinvite setting
(which I renamed) to actually be based on IP address masking, so that
Asterisk can make a more intelligent decision, but even that has
problems, because we don't actually _know_ that any given IP route
Spencer Nassar wrote:
Does anyone know if the tutorial materials from Atricon 2004 are
available for download anywhere? I'm particularly interested in Joachim
Vanheuverzwijn's Performance and Scalability tutorial slides (Asterisk -
building your system for performance and scalability).
Stig Andersson wrote:
So, I try
-
SetVar(cid=${CALLERIDNUM:-5:5})
The result is a empty string if CALLERIDNUM is less than 5 digits long,
which is NOT the case of SubString. SubString command returns what remains of
the variable,
that is - if CALLERIDNUM is 4 digits in length, it
During the last week, we have had several support issues being reported
as bugs on the bug tracker. Since we are going into a final development
stage on version 1.1dev (CVS HEAD) in order to complete the 1.2 release
we are under pressure to fix bugs and handle a lot of reports in a short
time
Anton Krall wrote:
I think it would be your last suggestion.. When I pickup the phone I hear a
tone, the sip phone box tone... Then I hit 9, no tones :) and enter the
whole phone number and it starts to ring on the other side.. So no outside
dialtone get heard ever.. I was wondering if it could be
Sarat Vemuri wrote:
While trying to deploy a bunch of Polycom IP 500 phones, I ran in to the
following. I limited the RTP ports from 8000-8050 to limit holes in
firewall. Pretty soon Asterisk ran out of RTP ports. Traced the
problem back to how * is handling SUBSCRIBE. A sip structure is
I've added an introduction article about the ARA on my web site
http://www.voip-forum.com/
The same text is now also added to CVS head as README.realtime.
On the same site, you will also find the news item about how we used
Asterisk for a call from an airline jet above Greenland to Stockholm,
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully
will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable
relase.
Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs
support in the form of funding in order to take the time to test this
Steve Underwood wrote:
re here: http://www.astertest.com/forum/viewtopic.php?t=13
Thank you for your contribution!
The hard work of building the thing was done for free, and now someone
brings out the begging bowl for the relatively minor activity or porting
into to another home. Frankly, that
In the Grandstream setup, turn off subscribe to message waiting
indication.
...or upgrade to CVS head, where I've fixed this problem with SUBSCRIBE.
Best regards,
/Olle
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Atif Rasheed wrote:
HI all,
I have the following setup running:
EP---Calling Asterisk---Relaying Asterisk---Softswitch--- PSTN
The Endpoint EP is registered with the Calling Asterisk. Calls are
forwarded from this machine to
Relaying Asterisk which in turn forwards it to the Softswitch. In
Jason T. Nelson wrote:
I have already started playing with trying to figure out why Asterisk runs
so badly under FreeBSD, such as eating 100% of the CPU without warning
unload pbx-wilcalu.so, see
http://www.voip-info.org/wiki-Asterisk+freebsd
/O
___
Rich Adamson wrote:
Has anyone played around with QoS or TOS relative to * and sip phones?
I was just doing a little real-time research and noticed our C7960's
mark IP packets with low delay and high throughput (presumably due
to tos_media: 5 in the SIPDefault config file), and rtp packets
Philipp von Klitzing wrote:
Hi there,
whenever I use a macro to dial out I see only s recorded in the dst
field of the CDR. Is there anyway to get around that problem except for
not using a macro?
Example:
)
Try to match every extension before dialing out instead, using s is a bad thing for
Dustin Goodwin wrote:
I did find something interesting. If you set reinvite=yes then * can
setup the RTP stream so that it avoids the media proxy in the * box
completely. I haven't tested to see if it changes anything.
Can we please kill reinvite - it does not exist in the SIP channel as an
John Todd wrote:
The soundfiles I submitted earlier today have been cleaned up, and added
to the Digium CVS server in a more formal manner. Also, some of the
really bad formatting in my .txt description file has been rectified.
All of the sounds on my website are now on the Digium site, and
Could you please explain what you want to do, why you want asterisk to register but
not take
the calls?
You could take the calls into the dialplan (extensions.conf) and dial out from there
with an agi
script that performed the same thing. If you have canreinvite=yes, asterisk will leave
the
Steven Critchfield wrote:
On Mon, 2004-01-19 at 05:19, Kim Hendrikse wrote:
Is there a search engine for this list?
Google
Use site:lists.digium.com to limit the search to just the list server.
...or http://search.voip-forum.com
Indexes our lists, the Wiki, asterisk.org and some related
[EMAIL PROTECTED] wrote:
Are you using the 0.7.1 tar distribution or CVS? I was able to compile
the 0.7.1 Asterisk program/sample config's to get a working system on a PC
with no sound device and no phone interfaces. This system is about as
simple as it can get (except for the 3 fixed disks in
T. Chan wrote:
I think what Todd was referring to was to JUST do the signaling proxy on the
Asterisk but not proxying the media.
This is the definition of a SIP proxy. Asterisk is a PBX that supports SIP, but
not really a SIP proxy. As a PBX, it wants to be in the middle of a call. As an
Tilghman Lesher wrote:
On Sunday 25 January 2004 11:19, Philipp von Klitzing wrote:
Jan 25 17:30:02 ERROR[40979]: chan_iax.c:4826 set_config: Unable to
load config iax1.conf
As a matter of chan_iax slowly moving towards the deprecated pile, to be
replaced everywhere with chan_iax2, chan_iax
mattf wrote:
Go ahead and edit the page. I've fixed several little errors on pages that I
didn't create. The voip-info.org Wiki is like the total-open-source Asterisk
manual. Although the total-access may be a problem in the future because all
someone has to do to delete everything is just to
Dmitry Mishchenko wrote:
Now major problems comes:
After starting asterisk it is trying to get all available CPU time.
I'm using standard config files. Turning off modules h323 and oss
didn't help.
Read
http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd
for advice on how to get rid
Steve Foy wrote:
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a call, the remote end just drops off. We're using software
SIP phones (SJPhone) connecting to * then out through analogue lines with
X100P cards.
Let's go through how SIP works in Asterisk compared with a SIP Proxy. Remember that
Asterisk is not designed to be a SIP Proxy, it's designed to be a Multi-VOIP and PSTN
PBX, a quite complicated task.
(I'm not going into all details (ACK, TRYING, RINGING etc))
We have two SIP users, Alice and
Brancaleoni Matteo wrote:
Hi.
If both Alice and Bob are connected without NAT, have the same codec support and have
canreinvite=yes
* Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it
goes directly from Alice to Bob
Not all UAs support a re-INVITE and in
Rich Adamson wrote:
So, what hardware or use is the SUBSCRIBE method used for in
chan_sip.c? I asked this question a while ago, and got resounding
silence. Maybe someone who is better at de-tangling C code than I am
could take a peek.
Not sure, but seems to me it came in about the time Olle
Florian Overkamp wrote:
Hi,
-Original Message-
So, what hardware or use is the SUBSCRIBE method used for in
chan_sip.c? I asked this question a while ago, and got
resounding silence. Maybe someone who is better at
de-tangling C code than I am could take a peek.
Hmm, dunno. Could
Florian Overkamp wrote:
Hi,
-Original Message-
So, what hardware or use is the SUBSCRIBE method used for in
chan_sip.c? I asked this question a while ago, and got
resounding silence. Maybe someone who is better at
de-tangling C code than I am could take a peek.
Hmm, dunno. Could
Rich Adamson wrote:
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
Debuuging SIP to a file:
asterisk -c | tee /tmp/sipdebug.log
then turn on 'sip debug' at the CLI
/O
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Would like to see a SIP debug
* The invite from the caller phone to Asterisk
* The invite from Asterisk to the called phone
As well as the configs (extensions.conf and sip.conf)
Can't reproduce in my servers.
/O
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[EMAIL
Vic Cross wrote:
On Sat, 7 Feb 2004, John Fraizer wrote:
snip all the trace data
Here are the configs:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 66.35.64.38 ; Address to bind to
context = default ;
Dustin Knuttgen wrote:
Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you.
Read http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
The problem is well-known.
/Olle
___
Tim Sailer wrote:
On Mon, Feb 09, 2004 at 01:37:55PM -0600, Eric Wieling wrote:
That's just the way Asterisk's dial command works.
Hmm. I see. If it can't create the channel for either reason
(busy or not registered), it's handled the same. I think I'll
kludge up a perl script to watch the SIP
Tim Sailer wrote:
I've looked, poked, and hoped, but I can't seem to make * understand
the difference between a SIP channel being busy or not being there.
Both come up as 'busy'. I would expect the unregistered SIP to be seen
as unavailable. Am I just missing something obvious, again?
I've heard
I'm going. Would be great to have an Asterisk gathering.
/O
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Jeff wrote:
1. Two SIP phones can login to FWD at the same time with the same
username/pwd , is this normal?
Ed explained this.
2. can Two SIP phones login to * at the same time with the same
username/pwd ? how to prevent this?
Well, a SIP proxy normally allows this. Asterisk is not a SIP proxy
Costa Tsaousis wrote:
I was trying to figure out all the valid options for a sip.conf and I
believe I found a few weird things (or just a few things that are weird
to me :) Anyway, I decided to post this here together with my questions
and notes in case other people need this info too or have
Costa Tsaousis wrote:
context= ; UP, the context name for placing calls
Q1: Why is there a context for peers?
We use peers in some other situations as well. This is strange and
rather undocumented, but an incoming call is first matched by
username with the defined users (including 'friends').
Fran Boon wrote:
On Sat, 2004-02-21 at 09:06, Costa Tsaousis wrote:
incominglimit= ; U- concurrent call limitations ( = 0 )
outgoinglimit= ; U- concurrent call limitations ( = 0 )
Q6: How is it possible for a type=user phone to have BOTH incoming and
outgoing limits?
Interesting question. Anyone
Costa Tsaousis wrote:
Sorry, I was on the wrong topic, canreinvite has
yes|no|update as keywords.
with UPDATE a SIP method UPDATE is initiatied to change the media path.
with YES, a new INVITE is issued within the current call. (a re-invite)
with NO, the call stays within asterisk.
Sorry for
Rich Adamson wrote:
How come * says 1010 is BUSY in the trace below? I would have guessed
UNAVAILABLE since 1010 is not logged on/registered.
Sounds right to me.
That's what has been programmed in the asterisk code and has been that way
since the beginning of time.
Is that right? I was afraid it
George Pajari wrote:
I am having trouble getting SIP phones to register with Asterisk. I know
that the phone can register with FWD and I have used tcpdump to see the
registration packets arrive at the Asterisk server, but nothing goes back.
How should I attack the problem?
What debugging tools
If you see nothing with full verbosity and SIP debug turned on, the Asterisk SIP channel gets nothing.
The reason why we always mix in NAT with questions like yours is that in 90% of the
cases, NAT
is the problem. It's just a standard response, like when Microsoft support tells you
to reinstall
Lars Fredriksson wrote:
Hi!
I'm trying to record som voiveprompts, and I've created a directory se in
/var/lib/asterisk/sounds - in that directory I've put files like
vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I
hear my own voice prompts!
But wehre should I place the
I noted that I have to put a load=res_parking.so before chan_capi.so in modules.conf,
since
chan_capi 3.1 uses some parking group stuff. Otherwise startup failed with error on
symbol
ast_get_group
Worth to notice in the README!
/O
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Stephen R. Besch wrote:
Sean Rodger wrote:
I'm using a grandstream phone with asterisk.
Everything seems to be working fine, but every once in a while talking to
someone, the call is dropped.
A loud busy signal immediately interrupts the call for the grandstream
user,
while the other person
Vic Cross wrote:
G'day Marc,
On Wed, 25 Feb 2004, Marc Fargas wrote:
Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info
(SP5002/S) and traed to register to asterisk, It seems to autentcate but
sniffing the net it shows a 407 proxy authen required error message and I
Seems like republica registers ok, but not republica2. Republica2 failes to authenticate.
You have a normal registration sequense here:
-Client sends a REGISTER without authentication
-Server sends trying...
-Server sends 407 Proxy auth (should be WWW auth) with challenge
-Clients ACK
-Client
Iain Stevenson wrote:
Works perfectly fine for me - but I'm not using rfc2683 - my Grandstream
uses the latest firmware and SIP INFO.
Iain
--On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt
[EMAIL PROTECTED] wrote:
I cannot get the Message Waiting Light (MWL) on my Grandstream phone
Stephen R. Besch wrote:
Olle E. Johansson wrote:
Going back to the subject, what does the grandstream really do,
SIP-wise, when you press
the transfer button?
4.3.7 Call Transfer The user can transfer an active call to a third
phone by using the Transfer button. The sequence is like
Going back to the subject, what does the grandstream really do, SIP-wise, when you
press
the transfer button?
/O
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Low, Adam wrote:
Hey All,
I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940 phones.
My issue is that from what I see in chan_sip.c there is no support for the
Remote-Party-ID field in relation to withholding the calling partys number.
This is a legal
Low, Adam wrote:
Could you please point me in direction of standard documents, drafts or documentation of this?
IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy.
Thank you for the pointer, as this is still a draft (a lot of SIP things are), it's
Fran Boon wrote:
Olle's chan_sip2 introduces a 3rd possibility:
Using templates autocreate peers for the majority of user options
storing just the passwords in the MYSQL database.
Combining this with MYSQL_FRIENDS, storing template= settings in a database
would be very powerful.
/O
Fran Boon wrote:
I guess I need to implement this with astdb instead of MySQL, since this
can be queried direct within the dialplan.
Would be lovely to have dbget/dbput routines for MySQL as well as just
db1!
Brian was working on odbcget/put. I think there's a beta uploaded on his
web site.
/O
Steve Beaumont wrote:
On the wiki pages it suggests that clients on the outside of NAT can connect
to an Asterisk server behind nat. (option no 3). The note suggests that this
can work with port forwarding and some 'header mangling magic'.
I have the port forwarding configured however, when I try
Senad Jordanovic wrote:
Hans-Henrik Andresen wrote:
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all
of them are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
Jean-Marc V. Liotier wrote:
On Sun, 2004-03-07 at 13:03, Philipp von Klitzing answered off-list:
[6040]
defaultip=192.168.1.40
Replace this with host=dynamic and see what happens.
That's it !
Thinking it was going to make things easier to diagnose, I had chosen to
set the phones with a static
Jean-Marc V. Liotier wrote:
On Mon, 2004-03-08 at 18:50, Olle E. Johansson wrote:
If you configure a static address, the PBX already know how to reach
the client and no registration is therefore needed (and not allowed in
asterisk).
Enabling registration makes the SIP device mobile across
Rich Adamson wrote:
exten = s,1,GotoIf(DBget(FEAT/ivron) == yes?bus-ivr-main|s|1)
Rich,
I haven't seen GotoIF calling another application, only $[] constructs.
See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20gotoif
for examples. Take the output of dbget into a variable and test
Check out the latest CVS, Mark applied changes to the code in this area tonight.
The rtp.c is changed, so the old patch in bugs.digium.com may not be necessary any
more.
/Olle
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Jon Lawrence wrote:
Hi,
I've got 2 x100p's installed in my system.
Both execute the same incoming contexts as follows:
[inboundA]
include = dialjon
[inboundB]
include = dialjon|09:00-16:30|Mon-Fri|*|*
[dialjon]
exten = s,1,answer
exten = s,2,Dial(SIP/2000,15)
exten = s,3,Playback(noone)
exten =
Jon Lawrence wrote:
Surely * should know if a phone is in use ? After all it initiated/took part
in the call in the first place ;)
Again, the SIP device is not a slave device. It could receive a call from
somewhere else and be busy without Asterisk having a clue. A lot of SIP UAs,
like Xten
Walker Haddock wrote:
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
The incominglimit limits how many simultaneous calls a UA may place to
Asterisk.
I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device. If you set incominglimit=1
Ignace CARIA wrote:
Hi,
How can I use my external Modem (US Robotics Sportster Flash - RS 232)
as a voice client connected to Asterisk.
You can't.
/Olle
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Ross Finlayson wrote:
voice mail. However, if I try to call user2 from user1's X-Lite -
or vice-versa - I get a 404 Not Found error.
Is there anything obvious that I'm doing wrong? (In particular, do I
also need to add entries to extensions.conf for user1 and user2??)
Ross.
Try
Mark Phillips wrote:
[EMAIL PROTECTED] said:
Is it Sip Registry Server ?.
Could it work as Proxy Server ?
Hello Ahmet,
Asterisk is more than a proxy. Its an entire PBX. At a basic level it can
be used as a proxy though.
My favourite subject... :-)
No, Asterisk is not even close to a SIP
Mark Spencer wrote:
The Asterisk community is growing at a remarkable pace. I know there are
thousands of you out there -- in fact there are over eight *thousand*
subscribers to asterisk-users alone, and almost one *thousand* registered
users on the bug tracker.
This means that everything
As I started this trend I take the right to end it.
I just want us to follow John Postel's rule for how to act on the Internet
(I think he defined it for TCP/IP software, but it can be applied here too.)
Be strict in what you send
Be generous in what you accept
Sending a reply to
Justin Carlson wrote:
I am sorry if this is a silly question but I can not seem to locate the
festival binaries. does this come with asterisk or is it another project?
No question is silly. This is a good time to remind the list of the FAQ
Thomas Gallaway wrote:
Here is my problem. I have 2 phones (Grandstream Budge Tone-100)
loosing the sip registration
every 4 hours. I can not find out why.
It seems like the registration fails, then a few minutes after
registers sucessfull.
Mar 19 14:06:14 NOTICE[147466]: Registration from
Fritz Müller wrote:
How can I configure * to store the caller and called Party IP Address in
the CDR file.
Depends on the channel, not all channels are IP based.
Check the CDRuserfield - it's a free field in the CDR you set in the
dialplan or from a script.
Without knowing why you want this, I
Senad Jordanovic wrote:
Does anyone know if qualify=XXX should be used ONLY for user agents
behind NAT.
No, you can use it to qualify any address. Qualification means that
Asterisk regurlarly sends SIP messages with the OPTION method and
the UA answers. We clock the time and if the client takes
Joao Carlos Moura wrote:
How can I settup a way for Asterisk doesn´t make any use of DIGEST
AUTHENTICATION method?
I don t want ASTERISK to check out any username or password of my users.
Set no secret in sip.conf our use autocreatepeer
/Olle
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CW_ASN wrote:
Try adding 'insecure=yes' in sip.conf.
insecure=yes doesn't help in regards to authentication, or?
Please explain more.
/O
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