Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Ciao
Just 2 remarks
the line:
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f0b840", "SIP/
1009/0144") in new stack
does not seem to come out of:
Exten => _1XXX,1,Dial(SIP/${EXTEN})
it should look like:
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f0b840", "SIP/
1009/1442302") in new stack
second
is the line cord plugged in and working?
Because
-- Got SIP response 503 "Service Unavailable" back from
192.168.0.111
is what you get when there is no line attached/working.
Ciao
Roberto
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Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hi Roberto,
i added this exntesions.conf
[spa]
Exten => _1XXX,1,Dial(SIP/${EXTEN})
exten => _0.,1,Dial(SIP/1009/${EXTEN:1})
and in sip.conf:
[1009]
username=1009
type=friend
secret=1234
host=dynamic
canreinvite=yes
context=spa
disallow=all
allow=alaw
dtmfmode=info
qualify=yes
callgroup=1
pickupgroup=1
which i have the extension 1009 in sip.conf directed to it..
and then tried calling out but it still gave me the same error!
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f0b840",
"SIP/1009/0144") in new stack
-- Called 1009/0144
-- Got SIP response 503 "Service Unavailable" back from 192.168.0.111
-- SIP/1009-0821d888 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1003-b5f0b840' status is 'CONGESTION'
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Wed, 21 May 2008 10:29:21 -0700
Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to
pstn calls)
Ciao Roland
your dialplan:Exten => _1XX,1,Dial(SIP/${EXTEN})
_1XX is a three (3) digit number starting with 1, I'm not sure what happens if
you dial 1009 but it seems that it is dialing.
Anyway the ${EXTEN} is 1009 so asterisk is trying to dial that extension which
doesn't exist.
your dial out should look something like:
[outgoing]
exten => _9.,1,Dial(SIP/100/${EXTEN:1})
where you're specifying that all the calls that starts with 9 will go to
extension 100 (assuming that is your spa-3102) and there you dial the number
dialed from the caller stripped by the 9 (that is the :1 after EXTEN)Now 9 is
standard in USA for outside line, in some other countries is 0, you choose
CiaoRoberto
On May 21, 2008, at 7:52 AM, RoLaNd RoLaNd wrote:
Hello Roberto,
first of all, id like to thank you for your help with this..
secondly, i tried the configuration you gave me but it still gave me the same
error..!
but just to b sure ill tell u wht im doing..
after following ur advice to the letter.. i kept my asterisk configuration the
same the only thing i edited in sip.conf is adding the port for the pstn
extension to match the one in sipura 3102.. and gave the PSTN line interface on
sipura the user id of " 1009"
then i called from my softphone 1009 so i could dial out..
and it gave me this error in asterisk cli:
Connect attempt from '127.0.0.1' unable to authenticate
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f0e828", "SIP/1009") in
new stack
-- Called 1009
-- Got SIP response 503 "Service Unavailable" back from 192.168.0.111
-- SIP/1009-0821d888 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1003-b5f0e828' status is 'CONGESTION'
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager.d/op-panel.conf': Found
== Parsing '/etc/asterisk/users.conf': Found
is that the right way of doing this?! do i call 1009 (pstn line user id) or
wht!
ps: could us hare with me ur sip.conf and extensions.conf please just to
compare mine with urs maybe something is missing!
once again thanks for ur help :)
> Message: 22
> Date: Wed, 21 May 2008 06:49:39 -0700
> From: Roberto Milani <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
> sip/sip to pstn calls)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="windows-1252"
>
> Hi Roland
>
> I have 2 linksys spa-3102 working pretty good both dialing in and out
> and I followed this instructions to set it up:
>
>
> update to the latest firmware then:
>
> ..Go to the first tab ?Voice? and sixth sub-tab ?Line 1?
> SIP Settings:
> ..SIP Port: Notice that it is set to 5060 for line 1 and 5061 for
> PSTN Line (next tab). These port values must be correctly transferred
> to the correct contexts in sip.conf.
> Proxy and registration:
> ..Proxy: 192.168.5.70 < The IP address of your Asterisk server
> Subscriber Information:
> ..Display Name: LivingRoom < This will be the test phone, but any
> name would do as lone as it is used in the configuration files.
> ..User ID: LivingRoom
> ..Password: SomePassword
> ..Auth ID: LivingRoom < probably not needed
> Dial Plan:
> ..Dial Plan: (*xx|[3469]11|0|00|[2-9]x|
> 1xxx[2-9]xS0|.) < We have 10 digit local dialing.
> The default is set for seven digit local dialing. Adjust as needed.
> ..Emergency Number: < Hmmm, I don?t know what to do here: it?s
> probably important, but it is poor form to dial 911 just
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Ciao Roland
your dialplan:
Exten => _1XX,1,Dial(SIP/${EXTEN})
_1XX is a three (3) digit number starting with 1, I'm not sure what
happens if you dial 1009 but it seems that it is dialing.
Anyway the ${EXTEN} is 1009 so asterisk is trying to dial that
extension which doesn't exist.
your dial out should look something like:
[outgoing]
exten => _9.,1,Dial(SIP/100/${EXTEN:1})
where you're specifying that all the calls that starts with 9 will go
to extension 100 (assuming that is your spa-3102) and there you dial
the number dialed from the caller stripped by the 9 (that is the :1
after EXTEN)
Now 9 is standard in USA for outside line, in some other countries is
0, you choose
Ciao
Roberto
On May 21, 2008, at 7:52 AM, RoLaNd RoLaNd wrote:
Hello Roberto,
first of all, id like to thank you for your help with this..
secondly, i tried the configuration you gave me but it still gave me
the same error..!
but just to b sure ill tell u wht im doing..
after following ur advice to the letter.. i kept my asterisk
configuration the same the only thing i edited in sip.conf is adding
the port for the pstn extension to match the one in sipura 3102..
and gave the PSTN line interface on sipura the user id of " 1009"
then i called from my softphone 1009 so i could dial out..
and it gave me this error in asterisk cli:
Connect attempt from '127.0.0.1' unable to authenticate
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f0e828", "SIP/1009")
in new stack
-- Called 1009
-- Got SIP response 503 "Service Unavailable" back from
192.168.0.111
-- SIP/1009-0821d888 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1003-b5f0e828' status is
'CONGESTION'
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager.d/op-panel.conf': Found
== Parsing '/etc/asterisk/users.conf': Found
is that the right way of doing this?! do i call 1009 (pstn line user
id) or wht!
ps: could us hare with me ur sip.conf and extensions.conf please
just to compare mine with urs maybe something is missing!
once again thanks for ur help :)
> Message: 22
> Date: Wed, 21 May 2008 06:49:39 -0700
> From: Roberto Milani <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
> sip/sip to pstn calls)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="windows-1252"
>
> Hi Roland
>
> I have 2 linksys spa-3102 working pretty good both dialing in and
out
> and I followed this instructions to set it up:
>
>
> update to the latest firmware then:
>
> ..Go to the first tab ?Voice? and sixth sub-tab ?Line 1?
> SIP Settings:
> ..SIP Port: Notice that it is set to 5060 for line 1 and 5061
for
> PSTN Line (next tab). These port values must be correctly
transferred
> to the correct contexts in sip.conf.
> Proxy and registration:
> ..Proxy: 192.168.5.70 < The IP address of your Asterisk server
> Subscriber Information:
> ..Display Name: LivingRoom < This will be the test phone, but
any
> name would do as lone as it is used in the configuration files.
> ..User ID: LivingRoom
> ..Password: SomePassword
> ..Auth ID: LivingRoom < probably not needed
> Dial Plan:
> ..Dial Plan: (*xx|[3469]11|0|00|[2-9]x|
> 1xxx[2-9]xS0|.) < We have 10 digit local
dialing.
> The default is set for seven digit local dialing. Adjust as needed.
> ..Emergency Number: < Hmmm, I don?t know what to do here: it?s
> probably important, but it is poor form to dial 911 just to
test. . .
> Help?
> Click Submit All Changes
>
> ..Go to the first tab ?Voice? and seventh sub-tab ?PSTN?:
> SIP Settings:
> ..SIP Port: Notice that it is set to 5061 for PSTN User and 5060
> for Line 1. These port values must be correctly transferred to the
> correct contexts in sip.conf.
> Proxy and Registration:
> ..Proxy: 192.168.5.70 < The IP address of your Asterisk server
> Subscriber Information:
> ..Display Name: PSTN1 < I have two lines so there is an PSTN2,
but
> we will not discuss it here.
> ..User ID: PSTN1
> ..Password: SomePassword
> ..Auth ID: PSTN1 < probably not needed.
> Dial Plans:
> ..Dial Plan 2: (S0<:PSTN1>) < That is an S-zero. The incoming
call
> will be passed to your extensions.conf file with extension ?PSTN1?
> where we will Playback a greeting to the caller and then playback
the
> main menu of our internal users and their extension numbers
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hi Jose, i just did that, doesnt seem to work.. its still giving me the same error Date: Wed, 21 May 2008 11:02:36 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls) I was seeing your print screen images, and the observation is. You are not doing any sip registration on the server since your Register option in the Tab PSTN Line is set to NO. you should change it to yes. (or add in the sip.conf the host=SPA_ip instead of dynamic). regards. -- Jose Flores Galicia <<[EMAIL PROTECTED]>> BriefCode && Code Based Training _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
I was seeing your print screen images, and the observation is. You are not doing any sip registration on the server since your Register option in the Tab PSTN Line is set to NO. you should change it to yes. (or add in the sip.conf the host=SPA_ip instead of dynamic). regards. -- Jose Flores Galicia <<[EMAIL PROTECTED]>> BriefCode && Code Based Training ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
yes thats the only thing i have in extensions.conf
should there be anything else?!
Message: 21Date: Wed, 21 May 2008 09:40:26 -0400From: Matt Watson <[EMAIL
PROTECTED]>Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
sip/sip to pstn calls)To: Asterisk Users Mailing List - Non-Commercial
Discussion Message-ID:<[EMAIL
PROTECTED]>Content-Type: text/plain; charset="us-ascii" Does your
extensions.conf have any more configuration than what you've shown? If not,
then you are lacking dialplan for anything but internal calls. --Matt From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNdSent:
Wednesday, May 21, 2008 9:01 AMTo: [EMAIL PROTECTED]: [asterisk-users] asterisk
and sipura 3102 (pstn to sip/sip to pstn calls) Hello all, its been a while im
trying to setup my asterisk/sipura 3102 to recieve/make calls from softphones
on pcs in my home..i've set up 5 SIP extensions in sip.conf and made the
dialing plan in extensions.conf..i could make calls from 1 sip phone to another
in my home.. but i cant call out using pstn line interface nor recieve
calls..please find below my topology as well as config info:
(192.168.0.0) LAN__ |
| |softphone asterisk
sipura-PSTN LINE Configuration: ASTERISK: sip.conf
[101]type=peerport=5062host=dynamicsecret=1234context=spa
[103]type=peerport=5061host=dynamicsecret=1234context=spa
[100]type=peerport=5061host=dynamicsecret=1234context=spa
[111]type=peerport=5060host=dynamicsecret=1234context=spa
== === EXTENSIONS.CONF
[spa]Exten => _1XX,1,Dial(SIP/${EXTEN})
== === and this is the
settings i have right now for sipura 3102 in my PSTN LINE:
http://img84.imageshack.us/my.php?image=40541922um2.jpg<http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg>
http://img98.imageshack.us/my.php?image=55448347ss9.jpg<http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg>
http://img262.imageshack.us/my.php?imag ...
472qz3.jpg<http://img262.imageshack.us/my.php?imag%20...%20472qz3.jpg> ps: i
read so many tutorials and none seems to help..lately whenever i try to call
out using my sipphone.. it gives me "503 service unavailable" and this is wht
shows on the CLI of asterisk when i set sip debug on..
ubuntu-pbx-desktop*CLI> == Connect attempt from '127.0.0.1' unable to
authenticate-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f05600",
"SIP/1009") in new stack-- Called 1009*CLI>-- Got SIP response 410
"Gone" back from 192.168.0.111-- SIP/1009-081741d0 is circuit-busy ==
Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel
'SIP/1003-b5f05600' status is 'CONGESTION'
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___
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To UNSUBSCRIBE or update options visit:
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[asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hello Roberto,
first of all, id like to thank you for your help with this..
secondly, i tried the configuration you gave me but it still gave me the same
error..!
but just to b sure ill tell u wht im doing..
after following ur advice to the letter.. i kept my asterisk configuration the
same the only thing i edited in sip.conf is adding the port for the pstn
extension to match the one in sipura 3102.. and gave the PSTN line interface on
sipura the user id of " 1009"
then i called from my softphone 1009 so i could dial out..
and it gave me this error in asterisk cli:
Connect attempt from '127.0.0.1' unable to authenticate-- Executing [EMAIL
PROTECTED]:1] Dial("SIP/1003-b5f0e828", "SIP/1009") in new stack-- Called
1009-- Got SIP response 503 "Service Unavailable" back from 192.168.0.111
-- SIP/1009-0821d888 is circuit-busy == Everyone is busy/congested at this
time (1:0/1/0) == Auto fallthrough, channel 'SIP/1003-b5f0e828' status is
'CONGESTION' == Parsing '/etc/asterisk/manager.conf': Found == Parsing
'/etc/asterisk/manager.d/op-panel.conf': Found == Parsing
'/etc/asterisk/users.conf': Found
is that the right way of doing this?! do i call 1009 (pstn line user id) or
wht!
ps: could us hare with me ur sip.conf and extensions.conf please just to
compare mine with urs maybe something is missing!
once again thanks for ur help :)
> Message: 22> Date: Wed, 21 May 2008 06:49:39 -0700> From: Roberto Milani
> <[EMAIL PROTECTED]>> Subject: Re: [asterisk-users] asterisk and sipura 3102
> (pstn to> sip/sip to pstn calls)> To: Asterisk Users Mailing List -
> Non-Commercial Discussion> > Message-ID:
> <[EMAIL PROTECTED]>> Content-Type: text/plain; charset="windows-1252"> > Hi
> Roland> > I have 2 linksys spa-3102 working pretty good both dialing in and
> out > and I followed this instructions to set it up:> > > update to the
> latest firmware then:> > ..Go to the first tab ?Voice? and sixth sub-tab
> ?Line 1?> SIP Settings:> ..SIP Port: Notice that it is set to 5060
> for line 1 and 5061 for > PSTN Line (next tab). These port values must be
> correctly transferred > to the correct contexts in sip.conf.> Proxy and
> registration:> ..Proxy: 192.168.5.70 < The IP address of your Asterisk
> server> Subscriber Information:> ..Display Name: LivingRoom < This
> will be the test phone, but any > name would do as lone as it is used in the
> configuration files.> ..User ID: LivingRoom> ..Password:
> SomePassword> ..Auth ID: LivingRoom < probably not needed> Dial
> Plan:> ..Dial Plan: (*xx|[3469]11|0|00|[2-9]x| >
> 1xxx[2-9]xS0|.) < We have 10 digit local dialing. > The
> default is set for seven digit local dialing. Adjust as needed.>
> ..Emergency Number: < Hmmm, I don?t know what to do here: it?s > probably
> important, but it is poor form to dial 911 just to test. . . > Help?>
> Click Submit All Changes> > ..Go to the first tab ?Voice? and seventh
> sub-tab ?PSTN?:> SIP Settings:> ..SIP Port: Notice that it is set to
> 5061 for PSTN User and 5060 > for Line 1. These port values must be correctly
> transferred to the > correct contexts in sip.conf.> Proxy and
> Registration:> ..Proxy: 192.168.5.70 < The IP address of your Asterisk
> server> Subscriber Information:> ..Display Name: PSTN1 < I have two
> lines so there is an PSTN2, but > we will not discuss it here.> ..User
> ID: PSTN1> ..Password: SomePassword> ..Auth ID: PSTN1 < probably not
> needed.> Dial Plans:> ..Dial Plan 2: (S0<:PSTN1>) < That is an
> S-zero. The incoming call > will be passed to your extensions.conf file with
> extension ?PSTN1? > where we will Playback a greeting to the caller and then
> playback the > main menu of our internal users and their extension numbers.
> You can > also use specific extension numbers, such as: (S0<:2091>), which
> will > send all incoming calls to that extension for processing. This might >
> work best with two or more external lines where a second call comes in >
> while the first is being processed through the main menu and extension >
> capture.> VoIP-To-PSTN Gateway Setup:> ..Line 1 VoIP Caller DP: 1 <
> Leave this at 1. The SPA3102 will use > the Dial Plan 1 (above = (xx.)) so
> all your Dial Plan decision making > will be done in the Asterisk
> extensions.conf file. The SPA3102 will > dial out whatever Asteris
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hi Roland
I have 2 linksys spa-3102 working pretty good both dialing in and out
and I followed this instructions to set it up:
update to the latest firmware then:
..Go to the first tab ‘Voice’ and sixth sub-tab ‘Line 1’
SIP Settings:
..SIP Port: Notice that it is set to 5060 for line 1 and 5061 for
PSTN Line (next tab). These port values must be correctly transferred
to the correct contexts in sip.conf.
Proxy and registration:
..Proxy: 192.168.5.70 < The IP address of your Asterisk server
Subscriber Information:
..Display Name: LivingRoom < This will be the test phone, but any
name would do as lone as it is used in the configuration files.
..User ID: LivingRoom
..Password: SomePassword
..Auth ID: LivingRoom < probably not needed
Dial Plan:
..Dial Plan: (*xx|[3469]11|0|00|[2-9]x|
1xxx[2-9]xS0|.) < We have 10 digit local dialing.
The default is set for seven digit local dialing. Adjust as needed.
..Emergency Number: < Hmmm, I don’t know what to do here: it’s
probably important, but it is poor form to dial 911 just to test. . .
Help?
Click Submit All Changes
..Go to the first tab ‘Voice’ and seventh sub-tab ‘PSTN’:
SIP Settings:
..SIP Port: Notice that it is set to 5061 for PSTN User and 5060
for Line 1. These port values must be correctly transferred to the
correct contexts in sip.conf.
Proxy and Registration:
..Proxy: 192.168.5.70 < The IP address of your Asterisk server
Subscriber Information:
..Display Name: PSTN1 < I have two lines so there is an PSTN2, but
we will not discuss it here.
..User ID: PSTN1
..Password: SomePassword
..Auth ID: PSTN1 < probably not needed.
Dial Plans:
..Dial Plan 2: (S0<:PSTN1>) < That is an S-zero. The incoming call
will be passed to your extensions.conf file with extension ‘PSTN1’
where we will Playback a greeting to the caller and then playback the
main menu of our internal users and their extension numbers. You can
also use specific extension numbers, such as: (S0<:2091>), which will
send all incoming calls to that extension for processing. This might
work best with two or more external lines where a second call comes in
while the first is being processed through the main menu and extension
capture.
VoIP-To-PSTN Gateway Setup:
..Line 1 VoIP Caller DP: 1 < Leave this at 1. The SPA3102 will use
the Dial Plan 1 (above = (xx.)) so all your Dial Plan decision making
will be done in the Asterisk extensions.conf file. The SPA3102 will
dial out whatever Asterisk hands out.
PSTN-To-VoIP Gateway Setup:
..PSTN Ring Thru Line 1: no < When this is ‘yes’, an incoming call
goes directly through to Line 1. We only want line 1 to ring when
Asterisk routs a call to it.
..PSTN CID for VoIP CID: yes < capture the Caller ID provided by
the incoming call and pass it through to Asterisk to display on your
internal phones.
..PSTN Caller Default DP: 2 < Change to 2. The incoming call will
be passed to your extensions.conf file with extension 's' as defined
in Dial Plan 2 (above).
..Off Hook While Calling VoIP: no < I read this in some Google
search. I don’t know what it does, but stuff seems to work. Help?
FXO Timer Values (sec):
..PSTN Answer Delay: 5 < Delay so that you can get the CID data.
NghtShd at http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html
claims that 5 seconds is long enough.
Click Submit All Changes
Ciao
Roberto
On May 21, 2008, at 6:00 AM, RoLaNd RoLaNd wrote:
Hello all,
its been a while im trying to setup my asterisk/sipura 3102 to
recieve/make calls from softphones on pcs in my home..
i've set up 5 SIP extensions in sip.conf and made the dialing plan
in extensions.conf..
i could make calls from 1 sip phone to another in my home.. but i
cant call out using pstn line interface nor recieve calls..
please find below my topology as well as config info:
(192.168.0.0)
LAN__
|| |
softphone asterisk sipura-PSTN LINE
Configuration:
ASTERISK:
sip.conf
[101]
type=peer
port=5062
host=dynamic
secret=1234
context=spa
[103]
type=peer
port=5061
host=dynamic
secret=1234
context=spa
[100]
type=peer
port=5061
host=dynamic
secret=1234
context=spa
[111]
type=peer
port=5060
host=dynamic
secret=1234
context=spa
== ===
EXTENSIONS.CONF
[spa]
Exten => _1XX,1,Dial(SIP/${EXTEN})
== ===
and this is the settings i have right now for sipura 3102 in my PSTN
LINE:
http://img84.imageshack.us/my.php?image=40541922um2.jpg
http://img98.imageshack.us/my.php?image=55448347ss9.jpg
http://img262.imageshack.us/my.php?imag ... 472qz3.jpg
ps: i read so many
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Does your extensions.conf have any more configuration than what you've shown?
If not, then you are lacking dialplan for anything but internal calls.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn
calls)
Hello all,
its been a while im trying to setup my asterisk/sipura 3102 to recieve/make
calls from softphones on pcs in my home..
i've set up 5 SIP extensions in sip.conf and made the dialing plan in
extensions.conf..
i could make calls from 1 sip phone to another in my home.. but i cant call out
using pstn line interface nor recieve calls..
please find below my topology as well as config info:
(192.168.0.0)
LAN__
|| |
softphone asterisk sipura-PSTN LINE
Configuration:
ASTERISK:
sip.conf
[101]
type=peer
port=5062
host=dynamic
secret=1234
context=spa
[103]
type=peer
port=5061
host=dynamic
secret=1234
context=spa
[100]
type=peer
port=5061
host=dynamic
secret=1234
context=spa
[111]
type=peer
port=5060
host=dynamic
secret=1234
context=spa
== ===
EXTENSIONS.CONF
[spa]
Exten => _1XX,1,Dial(SIP/${EXTEN})
== ===
and this is the settings i have right now for sipura 3102 in my PSTN LINE:
http://img84.imageshack.us/my.php?image=40541922um2.jpg<http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg>
http://img98.imageshack.us/my.php?image=55448347ss9.jpg<http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg>
http://img262.imageshack.us/my.php?imag ...
472qz3.jpg<http://img262.imageshack.us/my.php?imag%20...%20472qz3.jpg>
ps: i read so many tutorials and none seems to help..
lately whenever i try to call out using my sipphone.. it gives me "503 service
unavailable" and this is wht shows on the CLI of asterisk when i set sip debug
on..
ubuntu-pbx-desktop*CLI>
== Connect attempt from '127.0.0.1' unable to authenticate
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f05600", "SIP/1009") in
new stack
-- Called 1009*CLI>
-- Got SIP response 410 "Gone" back from 192.168.0.111
-- SIP/1009-081741d0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION'
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy! Try
it!<http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us>
___
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[asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hello all,
its been a while im trying to setup my asterisk/sipura 3102 to recieve/make
calls from softphones on pcs in my home..
i've set up 5 SIP extensions in sip.conf and made the dialing plan in
extensions.conf..
i could make calls from 1 sip phone to another in my home.. but i cant call out
using pstn line interface nor recieve calls..
please find below my topology as well as config info:
(192.168.0.0)
LAN__
|| |
softphone asterisk sipura-PSTN LINE
Configuration:
ASTERISK: sip.conf [101] type=peer port=5062 host=dynamic secret=1234
context=spa [103] type=peer port=5061 host=dynamic secret=1234 context=spa
[100] type=peer port=5061 host=dynamic secret=1234 context=spa [111] type=peer
port=5060 host=dynamic secret=1234 context=spa
== === EXTENSIONS.CONF
[spa] Exten => _1XX,1,Dial(SIP/${EXTEN})
== === and this is the
settings i have right now for sipura 3102 in my PSTN LINE:
http://img84.imageshack.us/my.php?image=40541922um2.jpg
http://img98.imageshack.us/my.php?image=55448347ss9.jpg
http://img262.imageshack.us/my.php?imag ... 472qz3.jpg
ps: i read so many tutorials and none seems to help..
lately whenever i try to call out using my sipphone.. it gives me "503 service
unavailable" and this is wht shows on the CLI of asterisk when i set sip debug
on..
ubuntu-pbx-desktop*CLI> == Connect attempt from '127.0.0.1' unable to
authenticate-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f05600",
"SIP/1009") in new stack-- Called 1009*CLI>-- Got SIP response 410
"Gone" back from 192.168.0.111-- SIP/1009-081741d0 is circuit-busy ==
Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel
'SIP/1003-b5f05600' status is 'CONGESTION'
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___
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