On Tue, Nov 12, 2019 at 3:06 AM Thomas Roos
wrote:
> Hi,
> when using some non dynamic host eg. host=192.168.111.153 in sip.conf
> asterisk is not considering specific peer options eg. directmedia=off,
> transport=tcp
> if I set host=dynamic and register the sip phone it works as expected.
> Is
Hi,
when using some non dynamic host eg. host=192.168.111.153 in sip.conf
asterisk is not considering specific peer options eg. directmedia=off,
transport=tcp
if I set host=dynamic and register the sip phone it works as expected.
Is this a bug or feature - I wanna disable the usage of
On Mon, Oct 27, 2014 at 6:35 PM, John Kiniston johnkinis...@gmail.com
wrote:
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
I'm using the copy
On Tue, Oct 28, 2014 at 9:38 AM, Kevin Harwell kharw...@digium.com wrote:
On Mon, Oct 27, 2014 at 6:35 PM, John Kiniston johnkinis...@gmail.com wrote:
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
I'm using the copy of the script that's included with Asterisk 13
Hi,
The dots in extension will work as special characters.
that's means in sip.conf and extensions.conf couldn't work if there's dot?
thank you
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Hi,
i want to ask about sip.conf extension.conf the configuration.
is it possibility to make sip.conf configuration like this
[1510891531543...@wlan.mnc089.mcc510.3gppnetwork.org]
type = friend
context = tutorial
username = 1510891531543...@wlan.mnc089.mcc510.3gppnetwork.org
secret = 12345
host
The dots in extension will work as special characters.
On 14/09/2014 8:06 pm, rafa alfurqan rafa.alfur...@gmail.com wrote:
Hi,
i want to ask about sip.conf extension.conf the configuration.
is it possibility to make sip.conf configuration like this
Hi,
On a 11.7.0 asterisk, I'm playing with timzone option.
When I'm setting this value to us or fr (as listed in indications.conf),
I'm still seeing this:
Language : us
Tonezone : Not set
Has someone a working example ?
Can you reproduce this ?
Regards
--
10 jul 2012 kl. 20:50 skrev Kevin P. Fleming:
On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well known
bug that if you have multiple interfaces with
6 jul 2012 kl. 23:18 skrev Felix Salfelder:
Hi there.
i am seriously stuck with an asterisk and sip problem.
the following sip.conf works with respect to some_peer:
[general]
bindaddr = x.y.z.w
nat = no
[some_peer]
type=peer
host=somehost
secret=somesecret
some other
On Tue, Jul 10, 2012 at 10:24:18AM +0200, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication.
Thanks for the reply.
in the meantime i've found a sort of workaround.
[general]
host = dynamic
; take some
On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well known
bug that if you have multiple interfaces with different IP networks,
Asterisk will send from the wrong
Hi there.
i am seriously stuck with an asterisk and sip problem.
the following sip.conf works with respect to some_peer:
[general]
bindaddr = x.y.z.w
nat = no
[some_peer]
type=peer
host=somehost
secret=somesecret
some other
unrelated options
here x.y.z.w is the ip address of the interface
Hi All,
I'm using Exchange as our voicemail system. Everything works fine until the
1 week mark when Exchange changes the port number used, then Asterisk 1.8
seg faults and I have no phones (unless I restart the U.M. service before
the 1 week period is up). Since that is a hack, I'm hoping
On Mon, Dec 27, 2010 at 3:38 AM, Olivier oza_4...@yahoo.fr wrote:
2010/12/26 Richard Kenner ken...@gnat.com
I'm confused exactly what's supported with LDAP and Asterisk. What I want
to do is to have SIP peer information read directly (in realtime) from
LDAP.
Can this be done? If so, with
2010/12/26 Richard Kenner ken...@gnat.com
I'm confused exactly what's supported with LDAP and Asterisk. What I want
to do is to have SIP peer information read directly (in realtime) from
LDAP.
Can this be done? If so, with what Asterisk versions?
I'm also a bit confused about what's
I'm confused exactly what's supported with LDAP and Asterisk. What I want
to do is to have SIP peer information read directly (in realtime) from LDAP.
Can this be done? If so, with what Asterisk versions?
--
_
-- Bandwidth and
On Sat, Dec 25, 2010 at 8:58 PM, Richard Kenner ken...@gnat.com wrote:
I'm confused exactly what's supported with LDAP and Asterisk. What I want
to do is to have SIP peer information read directly (in realtime) from LDAP.
Can this be done? If so, with what Asterisk versions?
--
Here is the
On 08/07/2010 01:11 AM, unsero...@aol.com wrote:
Why don't you use 'real' realtime meaning to have your sip peers in your
database?
Then you would not have to do a reload after adding new peers to your db.
And you can still have sip peers additionally in sip.conf.
I have all of my sip
Please can anyone help me with this ?!
I have tried renaming the sip.conf file, or tried including another file
into sip.conf like sippy.conf and then add sippy.conf =
mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working.
The only thing that changes something is my
You cannot use realtime static and the other realtime tables at the
same time. You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your database. Or use the realtime static table
for
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
You cannot use realtime static and the other realtime tables at the
same time. You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
Or use the realtime static table for everything.
What do you mean by everything ?! What is this everything ?!
You mean all the sip options in a database and so no sip.conf file ?!
Kind regards,
Jonas.
--
You cannot use realtime static and the other realtime tables at the
same time. You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your database. Or use the realtime static table
for
Hello list,
scrambling different pieces of info together I've come with the following :
I want to have my register = statements in a MySQL-database, so I've
made the following table.
table ast_config :
id 1
cat_metric 0
var_metric 0
commented 0
filename sip.conf
category general
Hi
In sip.conf, you generally have something like
[name]
..
username=
secret=
What is the difference between the name specified in brackets and the
username key ?
What the sip client should provide ?
What do we use in dialplan when trying to reach this client ?
--
On Tue, Jul 6, 2010 at 7:11 PM, Ruddy Gbaguidi plugwo...@micnes.com wrote:
What is the difference between the name specified in brackets and the
username key ?
Context and username.
What the sip client should provide ?
The client will tell you their settings
What do we use in dialplan when
On 02/19/10 08:54, Olle E. Johansson wrote:
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf)
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote:
You propably have a type=friend where the user part matches before you even
hit the peer part, where the insecure configuration parameter matches. There
is a confusion here on the From: username and the authentication
19 feb 2010 kl. 10.22 skrev Randy R:
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote:
You propably have a type=friend where the user part matches before you even
hit the peer part, where the insecure configuration parameter matches. There
is a confusion here on the
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf)
On 02/18/10 09:00, Olle E. Johansson wrote:
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf) files on the same asterisk server and with one set
on the
server.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu
Sent: Monday, January 25, 2010 7:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip.conf with versatel
Hi
My System is:
Asterisk 1.6 running on a Dell Server with two network interfaces.
eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has
the local ip 10.26.208.252
and the external ip 89.244.x.y
eth0 of the server is configured to 10.26.192.107
The Problem:
SIP registration
...@lists.digium.com] On Behalf Of Yves Arikoglu
Sent: Monday, January 25, 2010 7:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip.conf with versatel and two NICs very
strangeproblem
Hi
My System is:
Asterisk 1.6 running on a Dell Server with two network interfaces.
eth1
- Yves Arikoglu yves...@gmx.de wrote:
Hi
My System is:
Asterisk 1.6 running on a Dell Server with two network interfaces.
eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has
the local ip 10.26.208.252
and the external ip 89.244.x.y
Either a typo or you have an IP
thanx... a typo... the routers local ip is 10.26.208.253
yves
Tim Nelson schrieb:
- Yves Arikoglu yves...@gmx.de wrote:
Hi
My System is:
Asterisk 1.6 running on a Dell Server with two network interfaces.
eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has
the
- what's the difference between a subscribe request et a register
request ?
A subscription in the SIP protocol is saying Hey, I'd like to be
notified when something happens. This is most often used when a phone
wants to subscribe to the state of another extension, or to the status
of a
Hello
I have few questions :
- what's the difference between a subscribe request et a register request ?
- in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please
someone could explain how doest it work because I think i'm a little bit
confuse.
- if I configure a sip terminal in
Hello,
well let me explain one part of your question, the host parameter. if
you want to restrict the access to one ip you can say it here.
host=192.168.2.13 means, that you can only use this account from
192.168.0.13, eg. for security reasons. i recommend so set it to
dynamic at the moment and
On Wed, 2009-08-05 at 14:32 +0200, harry R wrote:
- what's the difference between a subscribe request et a register
request ?
A subscription in the SIP protocol is saying Hey, I'd like to be
notified when something happens. This is most often used when a phone
wants to subscribe to the state
I have the following set in sip.conf [general] section.
rtptimeout = 60
rtpholdtimeout = 300
I would like to set these to default, or null the general settings for one
upline friend as it is solely a fax peer (T38 over SIP)
How can this be easily done?
Michael
John Todd wrote:
Would it be so difficult to have perhaps two different proxies? One
would be for any SIP messages destined for IP addresses that were not
in any of the localnet= lines, and one would be for any SIP messages
destined for IP addresses that were destined for IP addresses
So, does anyone ever used outboundproxy in sip.conf with success?
Does it only send OUTBOUND calls via the proxy and not also internal
extension calls via that proxy?
Best Regards,
Ricardo.
___
-- Bandwidth and Colocation Provided by
Ricardo Carvalho wrote:
Does it only send OUTBOUND calls via the proxy and not also internal
extension calls via that proxy?
As has already been posted in your other threads about this subject,
Asterisk has no concept of an 'outbound' call at all. In that sense, the
name of this option in
Thanks Kevin.
Although it doesn't fit my needs, thanks for the explanation. I guess I'll
really have to combine Asterisk with OpenSer to do what I want.
Ricardo.
On Thu, Mar 26, 2009 at 1:07 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Ricardo Carvalho wrote:
Does it only send
On Mar 26, 2009, at 6:07 AM, Kevin P. Fleming wrote:
Ricardo Carvalho wrote:
Does it only send OUTBOUND calls via the proxy and not also internal
extension calls via that proxy?
As has already been posted in your other threads about this subject,
Asterisk has no concept of an 'outbound'
Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Tuesday, March 24, 2009 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip.conf outboundproxy
On 24 Mar
Hi,
I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of
Asterisk, but for the tests I made, every calls, even internal SIP calls
between extensions are sent over the proxy that I have specified with the
outboundproxy=xxx.xxx.xxx.xxx in sip.conf.
I think this isn't the
On 24 Mar 2009, at 17:51, Ricardo Carvalho wrote:
Hi,
I'm trying to enable sip.conf outboundproxy support in version
1.4.20.1 of Asterisk, but for the tests I made, every calls, even
internal SIP calls between extensions are sent over the proxy that I
have specified with the
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Tuesday, March 24, 2009 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip.conf outboundproxy
On 24 Mar
31 jan 2009 kl. 02.44 skrev Mike:
Replying to my own message. How difficult would it be to add a
bindaddr (and possibly bindport) PER PEER in SIP.conf?
How much of a bounty would I have to pay to get this done you think?
Well, if you run bindaddr=0.0.0.0 Asterisk will listen to all IP's.
hI,
Trying to understand how to setup two PRIs in sip.conf. Using Asterisk
1.4.23.
I have a provider giving me two PRI (different rate centers) through SIP.
Both PRI comes in from the same IP on the provider side, but go to two
different IPs (both on the same box) on my side.
How can I
30 jan 2009 kl. 16.59 skrev Mike:
hI,
Trying to understand how to setup two PRIs in sip.conf. Using
Asterisk 1.4.23.
I have a provider giving me two PRI (different rate centers) through
SIP. Both PRI comes in from the same IP on the provider side, but
go to two different IPs (both
] On Behalf Of Mike
Sent: Friday, January 30, 2009 10:59
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP.Conf - bindaddr per peer?
hI,
Trying to understand how to setup two PRIs in sip.conf. Using Asterisk
1.4.23.
I have a provider giving me
I currently have my phones setup in the sip.conf file. I use templates
to describe the specific settings to each phone type.
For instance in sip.conf, I have:
[generic_phone](!)
...
...
[polycom501](!,generic_phone)
...
...
[grandstream](!,generic_phone)
...
...
;begin subscribers
14 apr 2008 kl. 16.19 skrev Al lists:
I have seen this issue on both 1.2 and 1.4, was not able to
reproduce to find a cause or bug.
I have seen this after power failure boot up.
show sip peer command shows most of peers, except one or two (in my
cases trunk) .
if i issue a sip reload
I have seen this issue on both 1.2 and 1.4, was not able to reproduce to
find a cause or bug.
I have seen this after power failure boot up.
show sip peer command shows most of peers, except one or two (in my cases
trunk) .
if i issue a sip reload command, it will show all of them.
I can write a
Hi,
does anybody know about the setvar option in asterisk's sip.conf. I am
trying to define it for a peer that's used when making calls using the
originate ami call, but it seems to not have any effect.
Marcus
--
Marcus Hunger - [EMAIL PROTECTED]
Telefon: +49 (0)211-63 55 55-61
Telefax: +49
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like
So, wouldn't it be great to enable setvar for outgoing calls too?
On Fri, Mar 28, 2008 at 1:55 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in
On Fri, 2008-03-28 at 13:55 +0100, Johansson Olle E wrote:
Well, Jared, but that's the reverse. You stripped out this important
part:
am trying to define it for a peer that's used when making calls
using the originate ami call, but it seems to not have any effect.
The important thing
28 mar 2008 kl. 14.00 skrev Marcus Hunger:
So, wouldn't it be great to enable setvar for outgoing calls too?
Well, maybe in the outbound channel then. But that won't help much.
mixing the caller's and callee's variables in the INCOMING channel
would be messy and only cause issues.
But
Particularly, I want to set the SIPADDHEADER variable dynamicly for peers
with rt-engine. Working around it might be possible, but having the thing
working transparently for Dial and Originate would be great.
On Fri, Mar 28, 2008 at 2:47 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
28 mar
28 mar 2008 kl. 14.56 skrev Marcus Hunger:
Particularly, I want to set the SIPADDHEADER variable dynamicly for
peers with rt-engine. Working around it might be possible, but
having the thing working transparently for Dial and Originate would
be great.
That should work today with the
Ok,
Now I have a friday afternoon patch for the community.
In the branch
http://svn.digium.com/view/asterisk/team/oej/peer-chanvars/
there's an addition to the SIPPEER() dialplan function where you can
retrieve a setvar= channel variable defined in sip.conf for the peer.
The branch is based
First of all, if Asterisk is the client and it must register to the other side,
does the peer\user entry have to be in sip.conf, or can it be in ARA?
Second, why do all calls fall through to the last context specified, whether in
that peer\user definition or not? I'm assuming it's a typo
Subject: [asterisk-users] sip.conf help,inbound calls fall to last specified
context
First of all, if Asterisk is the client and it must register to the other
side, does the peer\user entry have to be in sip.conf, or can it be in ARA?
Second, why do all calls fall through to the last
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Saturday, December 29, 2007 12:10 AM
Subject: [asterisk-users] sip.conf realtime
Hi -
I'm looking into realtime and I'm having a bit of a problem with the SIP
part.
My review of the posts seems to indicate that I should use
Sent: Tuesday, December 25, 2007 1:14 AM
Subject: Re: [asterisk-users] sip.conf for internetcalls.com
Quoting Justin Case [EMAIL PROTECTED]:
What comes up in the Asterisk CLI?
When it's not working, nothing appears in the CLI even though I've used
set verbose 10.
Also it can be a NAT issue
Hi -
I'm looking into realtime and I'm having a bit of a problem with the SIP
part.
My review of the posts seems to indicate that I should use realtime static
for the [general] part of my sip.conf including the registration commands:
register=did:secret@domain/did context
and use realtime
Hi all,
Perhaps someone here could help me with this. I'm new to Asterisk, but
have already met with some success at getting my first system to work
with two different VoIP (SIP) providers: XS4ALL and InternetCalls.com.
The config
for the former works fine, but my InternetCalls.com config
What comes up in the Asterisk CLI ? Set debug and verbosity to 9 and see
what comes up. Also it can be a NAT issue ? Have Asterisk register every 3-4
minutes.
On Dec 24, 2007 4:00 PM, Jaap Winius [EMAIL PROTECTED] wrote:
Hi all,
Perhaps someone here could help me with this. I'm new to
Quoting Justin Case [EMAIL PROTECTED]:
What comes up in the Asterisk CLI?
When it's not working, nothing appears in the CLI even though I've used
set verbose 10.
Also it can be a NAT issue?
How can that lead to this intermittent behavior? I've already set
nat=yes. Also, I'm using an ADSL
We use the MAC of the phone (all lower case) with a -a, -b, -c, etc
tacked onto the end of the MAC to specify the line appearance.
One thing you MUST remember is that a sip.conf entry is NOT an
extension. Extensions are totally different from sip.conf entries.
sip.conf entries are DEVICES.
Erik Anderson wrote:
All - I've been wrestling with how to best structure the sip device
accounts on a new asterisk server I'm deploying. All of the sip
devices (currently only Linksys SPA941s) will reside on the same
subnet as the server, and I have already set up a decent automatic
All - I've been wrestling with how to best structure the sip device
accounts on a new asterisk server I'm deploying. All of the sip
devices (currently only Linksys SPA941s) will reside on the same
subnet as the server, and I have already set up a decent automatic
provisioning system for the
Use the extension, and use grep to determine which account uses which
phone. For example I provision my spa9xx phones from a subdirectory on
apache called spa which on slackware is at: /var/www/htdocs/spa/
doing:
grep 123 /var/www/htdocs/spa/* will tell you which phone it is.
On 9/18/07, Erik
Realtime and sip_buddies in mysql works well for very large
installations.
PaulH
On Tue, 2007-09-18 at 22:11 -0500, Erik Anderson wrote:
All - I've been wrestling with how to best structure the sip device
accounts on a new asterisk server I'm deploying. All of the sip
devices (currently
On 9/18/07, C F [EMAIL PROTECTED] wrote:
Use the extension, and use grep to determine which account uses which
phone. For example I provision my spa9xx phones from a subdirectory on
apache called spa which on slackware is at: /var/www/htdocs/spa/
doing:
grep 123 /var/www/htdocs/spa/* will
The obvious alternative is to use the extension as the sip UID:
Use the extension as the UID and add the mac address as a comment. Like so:
[123]
; Joe Smith
;mac=000E08DA0409
secret = blahblah
... and so on and so forth
This will give the best of both worlds. The mac is readily available and
Hi all,
I have some peers configured in SIP.CONF file with parameters
incominglimit and outgoinglimit set up to 10. By doing that, I expect
that this peer will not be allowed to handle more than 10 incoming calls
and 10 outgoing calls at the same time.
However, since I upgraded to Asterisk
Hi,
An observation on this feature, which I may have completely
misunderstood, so flame away if I am being dumb :)
Looking at the code, setting limitonpeers=yes causes all user and
peer calls to be ref-counted as if they are peer calls (assuming a
user and peer of the same name exist).
A
I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues
(Asterisk stops responding). I use VoIP Buster and in sip.conf I use
sip1.voipbuster.com. When I do sip show peers in CLI I get
voipbuster/tomo 194.221.62.207 5060 OK (27 ms)
And when I ping
Just curious how most of you are defining SIP peers in
sip.conf for Asterisk boxes talking to each other. Are most of
you just making a type=friend connection and a single context or are you
separating them out to in/out definitions and contexts?
In other words
Where voicegw1 is the
I use two user's per host one for user and the other peer. Sort of
like attahed.
I also prefer IAX for communication between asterisk boxes. IAX use's
less bandwidth than SIP and it's trunks are alot smaller. If you look
at SIP traffic, 80% of it is headers. The headers look just like smtp
-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] sip.conf for talking to other Asterisk
machines
I use two user's per host one for user and the other peer. Sort of
like attahed.
I also prefer IAX for communication between
Hi to all,
I am just trying for Asterisk on my fedora core3 machine. here's my sip extensions config files
sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
allow=all
context=ECPT
localnet=192.168.0.1
localmask=255.255.255.0
[phone1]
type=friend
host=192.168.0.53
context=Embedded
So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's
site at http://slacker.com/~nugget/projects/asterisk/page7
Wow, awesome, I can call anywhere now. However, I think there is a more
elegant way of figuring out whether or not the local * server should handle
a given
Hi,
When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw)
stop working and I get the frame type error for them, but g729 works fine.
I've cleared general part of sip.conf of codec info to be on safe side. If
ulaw and alaw are the only ones allowed they work fine. Asterisk
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network
can register with specific user?
The thing is that I can't use password and I can't use host=ip.of.my.phone. And
I have to be sure that no one, from Internet will register on my * like that
user.
So, please tell me
: [Asterisk-Users] Sip.conf
In sip.conf, how can I define that only IP phones from 192.168.0.0/24
network can register with specific user?
The thing is that I can't use password and I can't use host=ip.of.my.phone.
And I have to be sure that no one, from Internet will register on my * like
In sip.conf, how can I define that only IP phones from 192.168.0.0/24
network can register with specific user?
The thing is that I can't use password and I can't use
host=ip.of.my.phone. And I have to be sure that no one, from Internet
will register on my * like that user.
So, please
2006/4/18, Tomislav Parčina [EMAIL PROTECTED]:
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network
can register with specific user?
The thing is that I can't use password and I can't use host=ip.of.my.phone.
And I have to be sure that no one, from Internet will
Is there a document/wiki/web site that maps the various SIP.conf settings to
the structure of the actual IP packet?
If so please advise.
--
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http://www.voip-info.org/wiki-asterisk
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
On 12/8/05, John Voss [EMAIL PROTECTED] wrote:
Is there a document/wiki/web site that maps the various SIP.conf settings to
the structure of the actual IP packet?
If so please
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