Asterisk using Sipura SPA-3000 to place calls over pstn line.

[VoipUser]

I'm trying to get Asterisk to dail outboud using the SPA-3000 and the plain old telephone line attached to it. My configs below are cobbled together from various posts including this site. I am able to receive calls and have routed pstn to a sip phone I have connected on my desk. Just need to get it work the other way. Thanks in advance for looking this over.
output from sip show peers:

Name/username	Host	Dyn	Forcerport	Comedia	ACL	Port	Status	Description
700/700	192.168.7.40	D	Auto (No)	No		5060	Unmonitored
demo-bob	(Unspecified)		Auto (No)	No		0	Unmonitored
ptsn/ptsn	192.168.7.61	D	No	No		5061	Unmonitored
spa3000/spa3000	192.168.7.61	D	No	No		5060	Unmonitored

4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 1 offline]

The error I'm getting in the CLI:
Connected to Asterisk 17.7.0 currently running on pbx01 (pid = 3782)
== Using SIP RTP CoS mark 5
> 0x7f6c00007740 -- Strict RTP learning after remote address set to: 192.168.7.40:16626
-- Executing [96665555@local-phone:1] Goto("SIP/700-00000000", "outgoing,6576757") in new stack
-- Goto (local-phone,outgoing,6665555)
[Sep 17 15:26:42] WARNING[3842][C-00000001]: pbx.c:4510 __ast_pbx_run: Channel 'SIP/700-00000000' sent to invalid extension but no invalid handler: context,exten,priority=local-phone,outgoing,6665555

My sip.conf
[general]
tcpenable=yes
transport=udp
context=local-phone

[demo-bob]
type=friend
secret=somepwd2; put a strong, unique password here instead

[700]
type=friend
host=dynamic
secret=pass
disallow=all
allow=ulaw

[spa3000]
mailbox=3000 ; Mailbox number (numerical so can be selected through phone)
type=friend
canreinvite=yes
nat=no
host=dynamic
secret=someotherpwd
call-limit=2
disallow=all
allow=ulaw
allow=g729
allow=gsm

[ptsn]
type=friend
canreinvite=yes
host=dynamic
secret=somepwd
call-limit=2
nat=no
port=5061
disallow=all
allow=ulaw
;allow=g723.1
allow=g729
allow=gsm

My extensions.conf
[local-phone]
exten = 2002,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()

exten => 115,2,Dial(SIP/115) ; see “show application dial” for options and formats
exten => 115,4,Voicemail2(u115) ; go to Voicemail2 if phone is “U”nanswered
exten => 115,102,Voicemail2(b115) ; go to Voicemail2 if phone is “B”usy
exten => 115,103,Hangup ; and then hangup.

exten => 700,1, Dial(SIP/700,10,t)
exten => 700,4,Voicemail2(u700) ; go to Voicemail2 if phone is “U”nanswered
exten => 700,102,Voicemail2(b700) ; go to Voicemail2 if phone is “B”usy
exten => 700,103,Hangup ; and then hangup.

exten => _9X.,1,Goto(outgoing,${EXTEN:1})

[incoming]
exten => ptsn,1,Goto(local-phone,700,1)

[outgoing]
exten => _X.,1,Dial(SIP/ptsn,20)

 

[atsak]

Holy cow a Sipura 3000? Thats gotta be 15 years old.

I'm sorry I can't tell you how do this in raw asterisk. I use IncrediblePBX to generate all the configuration (which has a Freepbx derived front end to hand that). You might be better to try using something with a front end or ask this over at the Asterisk forums.

 

[VoipUser]

Holy cow a Sipura 3000? Thats gotta be 15 years old.

I'm sorry I can't tell you how do this in raw asterisk. I use IncrediblePBX to generate all the configuration (which has a Freepbx derived front end to hand that). You might be better to try using something with a front end or ask this over at the Asterisk forums.

Yes, I had to wipe the dust off it, but I'm told its been replaced with a linksys model that is almost an exact copy of it. Was hoping to not have to install Freepbx. Once I find the answer, I'll certainly post it.

 

[atsak]

The Linksys was also discontinued. There's a couple options from Patton, Grandstream and I think Sangoma now. I don't think Cisco's making something like it any more, but could be wrong.

 

[ostridge]

Search link Search for Cisco Linksys Sipura ATA SPA3000

This CISCO Analog Telephone Adapters Administration Guide may help.

I don't know how the pbx config dials via the FX0 port to PSTN.

A work around is to register and verify your PSTN number (with Sip Provider and provide the billing.PDF to prove ownership of pstn number. then you can use that number as your CLID outgoing thro your sip provider; unless the latest 'shake & stir' service changes have blocked that? That would need to send with P-Asserted Identity PAI and sendrpid=yes trustrpid=no or something?

Incidentally I have a SPA PAP2T using on line 1 to PAP2T port 49160 and Line2 to port 49260 to ensure different ports and away from causing confusion with 5160 (sip) and 5061 (pjsip) .

 

[VoipUser]

I finally got it working. I was trying to keep things in different contexts for added security. However, as I'm really just trying to use it as a bridge between a Cisco 7962 on my desk and my pots line, I ended up putting the Cisco and the SPA in the same context.

from my sip.conf
[ptsn]
type=friend
canreinvite=yes
host=dynamic
secret=somepassword
nat=no
port=5061

from my extensions.conf
exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@ptsn)

And on the SPA-3000 PTSN Tab subsection Dail plans
Dial Plan1 (<x. @gw0>)
Dial Plan2 (S0<:700>)

so any incoming call on the pots line gets forwarded to my desk sip phone which is extension 700