I appologize if this was already answered somwhere on http://www.voip-info.org/wiki-Asterisk, I'm sure it probably is. And if you wish to just point me to a link that would be appreciated. I am very new to asterisk and unix all around, so these questions may sound rather ignorant. First being, how do I setup asterisk to point to another asterisk server and make all the lines which should be PSTN or POTS go directly to another existing asterisk server by using accounts? For instance, if I was using another asterisk service with my voip phone to connect to it how could I make my local server use that account as a line? Also, Is there any really good documentation on configuring asterisk, besides the asterisk handbook. Maybe something a little more indepth and something explaining all the commands available on the console? Thanks a lot -chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040701/14c1ad33/attachment.htm
On Thu, 1 Jul 2004, chouck wrote:> First being, how do I setup asterisk to point to another asterisk server > and make all the lines which should be PSTN or POTS go directly to > another existing asterisk server by using accounts?You are looking for IAX - Inter Asterisk eXchange.> Also, Is there any really good documentation on configuring asterisk, > besides the asterisk handbook. Maybe something a little more indepth > and something explaining all the commands available on the console?Yes, google for "asterisk installation". Or "asterisk extensions.conf"... A few good points to start: http://www.automated.it/guidetoasterisk.htm http://users.pandora.be/Asterisk-PBX/InstallAsterisk.htm http://www.loligo.com/asterisk/ among many others...
Hi
I'm trying to put up an sip pbx system for my company but i'm getting
some
problems when I'm trying to call from server ( branch A ) to server ( branch
B ).
This is my extentions.conf :
exten => 3003,1,Dial,SIP/3003@192.168.0.200
________________________________________________________
And this is what I get when I try to dial that user in branch B
_________________________________________________________
-- Executing Dial("SIP/5001-66b1",
"SIP/3003@192.168.0.200") in new
stack
-- Called 3003@192.168.0.200
-- Got SIP response 404 "Not Found" back from 192.168.0.200
-- SIP/192.168.0.200-e638 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/5001-66b1' status is
'CONGESTION'
Both servers are exactly the same...
What can the problem be, that branch B server doesn't route the call through
Thx
Quintin
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If your looking to link 2 asterisk boxes might I suggest IAX. Much more
efficient in the way bandwidth
is utilized between the locations. Also if you want to use your sip
solution, have you setup the other
end point in your SIP.CONF? I have never got IP dialing to work in
asterisk but it works fine when
assigned in the conf file.
.o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Systems Engineer
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Quintin
Sent: Monday, 23 May, 2005 08:08
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] sip to sip
Hi
I'm trying to put up an sip pbx system for my company but i'm getting
some problems when I'm trying to call from server ( branch A ) to server
( branch B )...
This is my extentions.conf :
exten => 3003,1,Dial,SIP/3003@192.168.0.200
________________________________________________________
And this is what I get when I try to dial that user in branch B
_________________________________________________________
-- Executing Dial("SIP/5001-66b1",
"SIP/3003@192.168.0.200") in new
stack
-- Called 3003@192.168.0.200
-- Got SIP response 404 "Not Found" back from 192.168.0.200
-- SIP/192.168.0.200-e638 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/5001-66b1' status is
'CONGESTION'
Both servers are exactly the same.....
What can the problem be, that branch B server doesn't route the call
through
Thx
Quintin
This email was scanned by: Mcafee GroupShield
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Well yes and no. If they have static IP's then you only need to setup a
context as such:
You would assign the following information on your Branch B server with
BranchA's information.
[branchA]
type=friend
defaultip=xxx.xxx.xxx.xxx
context=default
insecure=yes
host=xxx.xxx.xxx.xxx
disallow=all
allow=g729
allow=alaw
allow=ulaw
You would do the same here but for the Branch A server with Branch B's
config.
[branchB]
type=friend
defaultip=xxx.xxx.xxx.xxx
context=default
insecure=yes
host=xxx.xxx.xxx.xxx
disallow=all
allow=g729
allow=alaw
allow=ulaw
In your extensions.conf your dialplan would look something like this:
exten => _30.,1,Dial(${EXTEN}@branchb,23,r) ; use this for calling
people on branch B
There is no need to register the boxes with each other if they are
static, which is the easiest way to set this up.
Any other questions lemme know..
.o-------------------------------------------------------o.
Brian Fertig
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Quintin
Sent: Monday, 23 May, 2005 09:23
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sip to sip
Hi B
Do you mean I must do this in my sip.conf file on eatch server
Branch A
register => 3001:1234@192.168.0.200 /3001
Branch B
register => 5001:1234@192.168.0.227 /5001
thx
Q
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian C.
Fertig
Sent: 23 May 2005 03:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] sip to sip
If your looking to link 2 asterisk boxes might I suggest IAX. Much more
efficient in the way bandwidth
is utilized between the locations. Also if you want to use your sip
solution, have you setup the other
end point in your SIP.CONF? I have never got IP dialing to work in
asterisk but it works fine when
assigned in the conf file.
.o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Systems Engineer
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Quintin
Sent: Monday, 23 May, 2005 08:08
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] sip to sip
Hi
I'm trying to put up an sip pbx system for my company but i'm getting
some problems when I'm trying to call from server ( branch A ) to server
( branch B )...
This is my extentions.conf :
exten => 3003,1,Dial,SIP/3003@192.168.0.200
________________________________________________________
And this is what I get when I try to dial that user in branch B
_________________________________________________________
-- Executing Dial("SIP/5001-66b1",
"SIP/3003@192.168.0.200") in new
stack
-- Called 3003@192.168.0.200
-- Got SIP response 404 "Not Found" back from 192.168.0.200
-- SIP/192.168.0.200-e638 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/5001-66b1' status is
'CONGESTION'
Both servers are exactly the same.....
What can the problem be, that branch B server doesn't route the call
through
Thx
Quintin
________________________________
This email was scanned by: Mcafee GroupShield
This email was scanned by: Mcafee GroupShield
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_____
From: Quintin [mailto:quintin@kulweb.co.za]
Sent: 23 May 2005 02:08 PM
To: 'asterisk-users@lists.digium.com'
Subject: sip to sip
Hi
I'm trying to put up an sip pbx system for my company but i'm getting
some
problems when I'm trying to call from server ( branch A ) to server ( branch
B ).
This is my extentions.conf :
exten => 3003,1,Dial,SIP/3003@192.168.0.200
________________________________________________________
And this is what I get when I try to dial that user in branch B
_________________________________________________________
-- Executing Dial("SIP/5001-66b1",
"SIP/3003@192.168.0.200") in new
stack
-- Called 3003@192.168.0.200
-- Got SIP response 404 "Not Found" back from 192.168.0.200
-- SIP/192.168.0.200-e638 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/5001-66b1' status is
'CONGESTION'
Both servers are exactly the same...
What can the problem be, that branch B server doesn't route the call through
Thx
Quintin
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