SPA3000 help

[mi_sanders]

Hi all,

For over a year, I have been playing with my SPA3000 off and on, and until a week ago when I tried the latest firmware 3.1.20, WAF was never there. Now I have no echo problem. Just a week ago, I put in pbxinaflash-1.1, updated FreePBX to 2.4.0, I left everything else alone. I tried the SPA3000 config from NV site, could not get for a life of me get the inbound PSTN work, I have virtually tried every configuration on Google I could find, I can not get any one of them work. So I am asking for help as last resort.

My Config: vbuzzer, regular bell PSTN line. I want to keep both as I would never have WAF on voip alone.
I want to control both from my PBX; I think that's a best way. I am open to ideas, suggestions, any sample config someone has that works that is similar to mine. Seriously I am tired of looking at it. I have reset everything more times than I can count.

Thanks in advance,
Mike.

 

[gbrook]

Before I offer my config which is for a 3102 but has worked forever what does WAF mean

 

[mi_sanders]

Before I offer my config which is for a 3102 but has worked forever what does WAF mean

WAF means "Wife Acceptance Factor"

 

[gbrook]

SPA 3102
PSTN Page (apart from normal stuff)
Subscriber Information
User ID: test_trunk
Dial Plans
Dial Plan 2 (S0<:YOUR PHONENUMBER)
PSTN-To-VoIP Gateway Setup
PSTN Caller Default DP: 2

FreePBX
Trunks
Trunk Name: test_trunk
PEER Details
username=test_trunk
type=peer
secret=YOURPASSWORDFROMSPA3102
qualify=yes
nat=no ;DEPENDS ON YOUR SITUATION
host=dynamic
dtmfmode=rfc2833 ;DEPENDS ON YOUR SITUATION
disallow=all
context=from-trunk
allow=alaw&ulaw ;DEPENDS ON YOUR SITUATION

USER Context:from_test_trunk
User Details
username=from_test_trunk
type=user
secret=YOURPASSWORDFROMSPA3102
qualify=yes
port=5061
nat=never ;DEPENDS ON YOUR SITUATION
host=THESPA3102IPADDRESS
dtmfmode=rfc2833 ;DEPENDS ON YOUR SITUATION
context=from-trunk
canreinvite=no

Inbound Call Control
DID: YOURPHONENUMBER
Send it to wherever you want

In sip_general_custom.conf
make sure that there is something like
; Allows Incoming SIP Calls e.g. test trunk
context=from-trunk

and that from-sip-external has been commented out in sip.conf

I assume that you have your sip_nat.conf set up as normal

Cheers
Garry
This has worked for me since day 1

 

[mi_sanders]

Thanx for the info gbrook , but it did not work.

I managed to get it to work using this document: http://isw.me.uk/content/view/14/28

Actually I used it last year, did not work, this time around worked. so far so good.

Mike.

 

[Q99]

SPA3000... SIP Trunks and ATA's, Oh my!

When getting started with A@H and a SPA3000, I followed Ward's setup of assigning the SPA3000 as an extension. Fortunately, it just didn't seem to sit quite right with me, so I went about getting it to configure as a trunk.

I published the steps under the old A@H wiki. I think it's probably still fairly similiar steps to get it accomplished. The old A@H wiki and my article is available at:

http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+5

Casey

 

[kcallis]

Still banging my head against the wall...

I have tried every suggestion, recommendation, passing thought, and clueless statements that have come my way to get my Linksys 3102 working and to no avail.

I have tried to stay away from creating an additional extension, but at this point I am willing to do what ever is necessary.

I previously had created an extension and some suggested that I not do that, and that hasn't worked either. I have moved passed the message that no channels are available to a simple "My call cannot be completed at this time"... I would like to believe that I am making some progress.

Any comments would be greatly appreciated!

 

[gbrook]

I assume that it is the FXO registration and working with PIAF and FreePBX that is the problem. If so I can try and help by providing my setup

Cheers
Garry

 

[grumpy]

Kim I too had many problems and searched for many days and tried it all.
What seemed to work was a combination of everything I read using bits and pieces. It now works at the very basic level with no bells or whistles. I will post my PIAF settings then check my SPA 3102 settings.

FreePBX difference from add new:

Extension
203
Display Name PSTN
secret XXXX

Trunk

Outbound Caller ID "Grumpy"<5555555555>
Maxium Channels 1
Dial Rules NXXXXXX

Trunk Name PSTN

username=PSTN
type=friend
secret=XXXX ;add yours from ata
qualify=yes
port=5061
nat=yes ; on local lan using hosted PBX
disallow=all
insecure=very
host=dynamic ;using hosted PBX ATA on dynamic ip locally if Asterisk is local use ip of ATA
context=from-pstn
allow=ulaw
canreinvite=no

Nothing under user or registration
Add trunk to routes

 

[grumpy]

ATA 3102

LINE 1 tab

Proxy and Registration

Proxy your asterisk ip
Use outbound proxy no
Register yes
Use OB proxy yes ; not sure why as there is nothing in there
Make call without reg yes
Ans cakk without reg yes

Subscriber Information

Display name spa3k phone
password same as in asterisk
user id 203

This is the dial plan which I believe is the default
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

PSTN Line tab

Sip port is 5061

Proxy your asterisk ip
Register yes
Use OB proxy yes ; not sure why as there is nothing in there
Make call without reg yes
Ans cakk without reg yes

Subscriber Information

Display name spa3k phone
password same as in asterisk
user id PSTN
Auth ID PSTN ; not used I think
Use Auth ID No

Dial Plans

Dial Plan 1: (xx.)
Dial Plan 2 (S0<:s>)
The rest same as 1

VoIP-To-PSTN Gateway Setup

VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: HTTP Digest
no dialing plans

VoIP Users and Passwords (HTTP Authentication)

VoIP User 1 Auth ID: PSTN
VoIP User 1 Password: same as asterisk PSTN pass
VoIP User 1 DP: none

PSTN-To-VoIP Gateway Setup

PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: yes PSTN PIN Max Retry: 3
PSTN CID For VoIP CID: yes PSTN CID Number Prefix:
PSTN Caller Default DP: 2

Regional tab the only things I changed thier was date and time and time zone

Router tab
Wan setup I changed to DHCP, gave it a static ip
Lan setup Bridge, disabled DHCP,

These are the settings that I remembered changing, on the info page it shows line 1 as registered and pstn as registered

Good luck.

If it doesn't work, I could email you my spa configuration if I can figure out how to do it with scrapbook for Firefox.

If your using DSL use a line filter on the PSTN ine in at the SPA 3102. (I found out the hard way, worked alright when server was local but when I went to hosted it crapped out the DSL connection)

 

[kcallis]

Where you mention PSTN

Don't you mean the extension assigned as opposed to PSTN? As it stands now, I am getting a failed registration on the PSTN side of the 3102.

ATA 3102



Subscriber Information

Display name spa3k phone
password same as in asterisk
user id PSTN
Auth ID PSTN ; not used I think
Use Auth ID No

 

[grumpy]

PSTN is the name of the trunk in my example, and the password is defined in the trunk details.
In the ATA 3102 tabs
line 1 = extension
PSTN = trunk.

 

[Hat]

Greetings all,

FWIW, this is how I was able to get my Spa 3000 to work with PIAF.

SPA 3000

System Tab

Internet Connection Type

• DHCP: no
• Static IP: xxx.xxx.xxx.xxx (whatever works with your network)
• Netmask: match your network
• Gateway: match you network

SIP Tab
RTP Packet Size: .020 (from the default of .030)

LINE 1 Tab
Proxy and Registration

• Proxy: PIAF IP
• Use Outbound Proxy: no
• Register: yes
• Use OB Proxy: yes
• Make Call Without Reg: no (seems to me that if this is set to yes, it would defeat the purpose of the Proxy registration)
• Ans Call Without Reg: no (see above)

Subscriber Information

• Display Name: (I left this blank)
• User ID: 203 (whatever extension you have created)
• Password: (whatever you used for the above extension)
• Use Auth ID: no (default setting)

Voip Fallback to PSTN: Yes

Dial Plan
I used the default dial plan. Though I wonder if this is even used because of the PIAF outbound route dialplan.



PSTN Line Tab

Sip Settings

• Sip Port: 5061 (may already be the default setting)

Proxy and Registration

• Proxy: PIAF IP
• Register: yes
• Make Call Without Reg: no
• Ans Call Without Reg: no

Subscriber Information

• Display name: I left this blank
• User ID: same as the trunk name in PIAF
• Password: same as the trunk in PIAF
• Use Auth ID: No
• Auth ID PSTN: blank

Dial Plans

• Dial Plan 2 (S0<:same as Inbound Route Description>)

VOIP-To-PSTN Gateway Setup

• VoIP-To-PSTN Gateway Enable: yes
• VoIP Caller Auth Method: HTTP Digest
• One Stage Dialing: yes
• Line 1 VoIP Caller DP: None
• VoIP Caller Default DP: None
• Line 1 Fallback DP: None

VoIP Users and Passwords (HTTP Authentication)

• VoIP User 1 Auth ID: same as the trunk name
• VoIP User 1 Password: same as trunk password in PIAF
• VoIP User 1 DP: none

PSTN-To-VoIP Gateway Setup

• PSTN-To-VoIP Gateway Enable: yes
• PSTN Caller Auth Method: none
• PSTN Ring Thru Line 1: no
• PSTN PIN Max Retry: 3
• PSTN CID For VoIP CID: yes
• PSTN CID Number Prefix: I left this field empty
• PSTN Caller Default DP: 2

FXO Timer Values (sec)

• VoIP Answer Delay: 1
• PSTN Answer Delay: 3 (if you have caller ID, if not, then 1 would work)
• I did not change any other values

PIAF/FreePBX

Create a SIP Trunk

Outbound Caller ID: <whatever you want>
Maximum Channels: 1
Trunk Name: Must be the same as was input in the User ID of the PSTN Tab in the SPA 3000
Peer Details
canreinvite=no
secret=must match the password of the PSTN Tab in the SPA 3000.
type=peer
qualify=yes
host=dynamic (I had the IP of the SPA here and it would not register. Not realy sure why.
nat=no
dtmfmode=rfc2833
context=from-pstn (I tried from-trunk and it would not register.)
disallow=all
allow=alaw&ulaw (or whatever works for you)

Once you create this trunk, you should place it at the top of the list of all trunks. This was the only way I was able to get the outbound route to use this trunk.

Create an Outbound Route

Route Name: Whatever you want
Route Password: Leave this blank
Dial Plan: Choose what ever pattern you want to go out on your POTS line.
Trunk Sequence: Must place the trunk created above as the first one.

Create an Inbound Route

Description: Must match the Dial Plan 2 entry in the SPA
DID Number: I left this blank
Caller ID Number: I left this blank
Set Destination: Whatever you want. I set it to a Ring Group, so it rings in like a plain phone call would

Create an Extension

User Extension: 203 (must match the User ID in the Line 1 Tab of the SPA
Display Name: whatever you want
Secret: must match the Password in the Line 1 Tab of the SPA

I am running a SPA 3000 with firmware 3.1.10(GWd).

Hope this helps

Tom

 

[grumpy]

Tom looks almost the same as mine which works fine, but I do have a question.

In the PIAF section you did not show the creation of an extension, instead you show an incoming route, why?

My setup currently has a all DID/any CID, with the extension 203 as the PSTN.

Maybe Ward can add a link to this from his SPA setup document. I spent many hours trying different things that did not work, I know this is just a basic setup but it does work.

I wonder if Kim's works now as I haven't heard from her?

 

[Hat]

Tom looks almost the same as mine which works fine, but I do have a question.

In the PIAF section you did not show the creation of an extension, instead you show an incoming route, why?

Grumpy,

I did follow yours and bits and pieces of others. The major problems that I had were in the trunk details. In particular the Host and Context.

In regards to the extenson, I forgot to mention that I did set one up in PIAF. I did mention an extension in the SPA setup, Line 1. This was setup just to use a standard phone. Things tend to slip when posting at 3 AM.

 

[grumpy]

context=from-pstn

I tried others too that did not work, I'm assuming that there is a default dialing plan for this to work.

As for host the PIAF IP on a local install worked for me but on the hosted PIAF I had to change to dynamic. Maybe it has something to do with DNS which I had running on my router.
If you add your steps for doing the extension it will be a complete how-to for anyone setting up the SPA 3102. Written a lot better than my example, I'm sure it will help many when they find it.

 

[Hat]

Grumpy,

I added the extension portion in my previous post. Also, thanks for your info, it went a long way in helping me get this to work.

Tom

 

[kcallis]

Still plodding long

Tom looks almost the same as mine which works fine, but I do have a question.

In the PIAF section you did not show the creation of an extension, instead you show an incoming route, why?

My setup currently has a all DID/any CID, with the extension 203 as the PSTN.

Maybe Ward can add a link to this from his SPA setup document. I spent many hours trying different things that did not work, I know this is just a basic setup but it does work.

I wonder if Kim's works now as I haven't heard from her?

Actually, her is actually him, but no offense taken! I has just decided to take a break of a couple of days since I had more pressing matters. Of course, the 3102 is nicely registered, but I am still not able to dial out or for that matter receive calls via PSTN

I created an extension on Line 1 pages (it is currently 500) and I created another extension (599) which I thought I was suppose to use in conjunction with the PSTN side of things. I can pick up my phone and dial with-in the box to other extensions all day long, but I get absolutely nothing except "All circuits are busy right now"...

I tried to make use of the outbound 9_Outside to access PSTN

The Dial rule is real simple:
9NXXNXXXXXX
9NXXNXXX
9NXX


And on the trunk side the dial rules

9|.

Although I never add an outbound dial prefix to the trunk. So once again, close but no cigar!

K.

 

[grumpy]

Kim, sorry about that gender thing; you know how ASSumptions work.

May I suggest that you change your dial rules to a even simpler form,

Trunk NXXXXXX or just a .
route NXXXXXX

Just to see if it works, move the PSTN route to the top so it is the first choice of routes. In putty asterisk -rvvvv to watch what is happening when you try to dial out. "All circuits are busy right now"... At least lets you know your getting to asterisk with the ATA

 

[kcallis]

Dial rules

The reason I tried to utilized 9 was because I have other ITSP in play. On the other hand, considering that it is free long distance on the PSTN, I could just do simple rules across the board.
What I am currently getting when I dial out is:

-- Executing [916195157234@from-internal:1] Macro("SIP/500-09abcc80", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/500-09abcc80", "user-callerid: device 500") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/500-09abcc80", "AMPUSER=500") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/500-09abcc80", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/500-09abcc80", "1|Set|REALCALLERIDNUM=500") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/500-09abcc80", "REALCALLERIDNUM is 500") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/500-09abcc80", "AMPUSER=500") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/500-09abcc80", "AMPUSERCIDNAME=House") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/500-09abcc80", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/500-09abcc80", "AMPUSERCID=500") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/500-09abcc80", "CALLERID(all)="House" <500>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/500-09abcc80", "REALCALLERIDNUM=500") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/500-09abcc80", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/500-09abcc80", "TTL: ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/500-09abcc80", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/500-09abcc80", "Using CallerID "House" <500>") in new stack
-- Executing [916195157234@from-internal:2] Set("SIP/500-09abcc80", "_NODEST=") in new stack
-- Executing [916195157234@from-internal:3] Macro("SIP/500-09abcc80", "record-enable|500|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/500-09abcc80", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/500-09abcc80", "recordingcheck|20080419-194459|1208659499.0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080419-194459|1208659499.0: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/500-09abcc80", "No recording needed") in new stack
-- Executing [916195157234@from-internal:4] Macro("SIP/500-09abcc80", "dialout-trunk|11|916195157234||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/500-09abcc80", "DIAL_TRUNK=11") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/500-09abcc80", "0|Authenticate|") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/500-09abcc80", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/500-09abcc80", "DIAL_NUMBER=916195157234") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/500-09abcc80", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/500-09abcc80", "GROUP()=OUT_11") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/500-09abcc80", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/500-09abcc80", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/500-09abcc80", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/500-09abcc80", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/500-09abcc80", "outbound-callerid|11") in new stack
-- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/500-09abcc80", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/500-09abcc80", "REALCALLERIDNUM is 500") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/500-09abcc80", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s@macro-outbound-callerid:9] Set("SIP/500-09abcc80", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:10] Set("SIP/500-09abcc80", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/500-09abcc80", "TRUNKOUTCID="K. Callis" <9282195181>") in new stack
-- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/500-09abcc80", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/500-09abcc80", "0?usercid") in new stack
-- Executing [s@macro-outbound-callerid:17] Set("SIP/500-09abcc80", "CALLERID(all)=K. Callis <9282195181>") in new stack
-- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/500-09abcc80", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s@macro-outbound-callerid:22] NoOp("SIP/500-09abcc80", "CallerID set to "K. Callis" <9282195181>") in new stack
-- Executing [s@macro-dialout-trunk:12] AGI("SIP/500-09abcc80", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 9|.
== fixlocalprefix: Dialpattern 9|. matched. 916195157234 -> 16195157234
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/500-09abcc80", "OUTNUM=16195157234") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/500-09abcc80", "custom=SIP/PSTN") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/500-09abcc80", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/500-09abcc80", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/500-09abcc80", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/500-09abcc80", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:20] Dial("SIP/500-09abcc80", "SIP/PSTN/16195157234|300|") in new stack
-- Couldn't call PSTN/16195157234
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/500-09abcc80", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/500-09abcc80", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/500-09abcc80", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing [916195157234@from-internal:5] Macro("SIP/500-09abcc80", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/500-09abcc80", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/500-09abcc80> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/500-09abcc80", "pls-try-call-later|noanswer") in new stack
-- <SIP/500-09abcc80> Playing 'pls-try-call-later' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/500-09abcc80", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/500-09abcc80", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/500-09abcc80", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/500-09abcc80", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/500-09abcc80", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/500-09abcc80", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/500-09abcc80", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/500-09abcc80' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/500-09abcc80' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/500-09abcc80'

pbx*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
PSTN/PSTN 192.168.1.150 D 5071 OK (8 ms)
500/500 192.168.1.150 D N 5070 OK (8 ms)
599/599 192.168.1.150 5071 OK (8 ms)
18 sip peers [Monitored: 13 online, 5 offline Unmonitored: 0 online, 0 offline]