asterisk users - May 2008 - 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 [1009 at spa: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'
 
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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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of RoLaNd RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To: asterisk-users at 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 [1009 at spa: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'


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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]xxxxxxxxx| 
1xxx[2-9]xxxxxxxxxS0|xxxxxxxxxxxx.) < 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 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 [1009 at spa: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! 
> _______________________________________________
> -- 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
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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
[1009 at spa: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 <roberto.milani at sbcglobal.net>> Subject: Re: [asterisk-users]
asterisk and sipura 3102 (pstn to> sip/sip to pstn calls)> To: Asterisk
Users Mailing List - Non-Commercial Discussion> <asterisk-users at
lists.digium.com>> Message-ID: <D01A8127-5C23-4329-8A5A-4079203B0B99 at
sbcglobal.net>> 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]xxxxxxxxx| >
1xxx[2-9]xxxxxxxxxS0|xxxxxxxxxxxx.) < 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 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 [1009 at spa: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! > > _______________________________________________> >
-- 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> > --------------
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> > ------------------------------> >
_______________________________________________> --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> > End of
asterisk-users Digest, Vol 46, Issue 69>
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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 <mwatson at
becon.org>Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
sip/sip to pstn	calls)To: Asterisk Users Mailing List - Non-Commercial
Discussion	<asterisk-users at lists.digium.com>Message-ID:
<60BDBA04C2769C49AD1416789D9123A00F9884C9F8 at
columbia.becon.int>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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of RoLaNd
RoLaNdSent: Wednesday, May 21, 2008 9:01 AMTo: asterisk-users at
lists.digium.comSubject: [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 [1009 at spa: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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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
<<FloJoSe at gmail.com>>
BriefCode && Code Based Training
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