Minimum implementation of Asterisk for Google Voice?

I have successfully installed FreePBX/Asterisk on Dockstar with Google Voice integration based on PlugPBX board. I have been use it as home server for GV dial out for many weeks now. It is very reliable. It consumes little power and a dedicated PBX is really very desirable.
Now considering my needs, installing full FreePBX is really overkill. I was thinking if it can simplified.
I installed Asterisk using apt-get successfully. However, since I am not familiar with Asterisk programming, I just copied a few key files (extensions*.conf, features*.conf and sip*.conf) from the FreePBX installation and it works well (with python scripts installed). The sip and features files are really simple. I feel there are a lot of things I don't need in the extensions files but I don't know how to remove them. I don't know if anyone has a proven minimum extensions.conf that can work with GV and I am interested to trying out.

I solved this problem and the results are posted here.
Please let me know your comments.

For the last few years I have been using a Linksys PAP2 adapter to connect a regular phone to Google Voice/Gizmo5 for free US calls and its been great. I just received notice from Google that the Gizmo service will be discontinued on April 3rd so its time to find a replacement.

twinclouds, do you know if the current asterisk in the debian squeeze arm repos works on squeeze and would it work using a PAP2 adapter rather than a SPA3102?

Alternatively I may go the OBi110 route if I can find one at a reasonable price.

-PG

twinclouds, I see your instructions for asterisk 1.8 at plugpbx.org - is this working good now?

-PG

Wow! Asterisk 1.8.3 works on Squeeze and no need for Gizmo or SIP numbers etc. :-)

Quick How To:

1) Follow twinclouds excellent instructions

2) overlay the google voice config from superm1

mkdir /etc/asterisk/tmp
cd /etc/asterisk/tmp
wget -r -nd http://www.arctangent.net/~superm1/gv_configs/
cd /etc
tar cvfz asterisk_config.tgz asterisk # backup - just in case :-)
cd /etc/asterisk
cp /etc/asterisk/tmp/*.conf /etc/asterisk

3) modify the config for gmail address peter.gunn@gmail.com with password peters_password

perl -pi -e 's/Mario Limonciello/Peter Gunn/g' /etc/asterisk/sip.conf
perl -pi -e 's/superm1/peter.gunn/g' /etc/asterisk/sip.conf /etc/asterisk/jabber.conf /etc/asterisk/extensions.conf /etc/asterisk/gtalk.conf
perl -pi -e 's/secret=.*/secret=peters_password/g' /etc/asterisk/jabber.conf

4) log into http://voice.google.com, goto Settings->Phones, and make sure "Google Chat" is selected

5) restart asterisk

/etc/init.d/asterisk restart

6) Configure your VOIP Phone or ATA adapter

My settings for the Linksys PAP2...
SIP Port: 5060
Proxy: 192.168.1.7 (IP address of your dockstar running asterisk)
Display Name: Peter (anything works here)
UserID: 101
Password:  (blank)
AuthID: 101
User AuthID: No

And thats it - all calls to my Google Voice phone number ring the phone plugged into the Linksys PAP2 adapter and any outgoing calls go out as if I dialed them on using Google Voice.

No accounts/fees, no SIP numbers/proxies/DIDs/etc., no opening/forwarding ports on my firewall - just works like it should!

Awesome! :-)

-PG



Edited 1 time(s). Last edit at 03/22/2011 09:07PM by petergunn.

The following minimal asterisk modules.conf allows Google Voice connectivity and dramatically reduces memory usage down to ~6Mb (original was 70+Mb!):

/etc/asterisk/modules.conf

[modules]
autoload=no

load => res_rtp_asterisk.so
load => res_jabber.so
load => chan_gtalk.so
load=res_crypto.so             ; Cryptographic Digital Signatures         
load=cdr_csv.so                ; Comma Separated Values CDR Backend       
load=chan_sip.so               ; Session Initiation Protocol (SIP)       
load=codec_g726.so             ; ITU G.726-32kbps G726 Transcoder         
load=codec_gsm.so              ; GSM/PCM16 (signed linear) Codec Translat
load=codec_ulaw.so             ; Mu-law Coder/Decoder                     
load=format_gsm.so             ; Raw GSM data                             
load=format_ilbc.so            ; Raw iLBC data                           
load=format_pcm.so             ; Raw uLaw 8khz Audio support (PCM)       
load=format_sln.so             ; Raw Signed Linear Audio support (SLN)   
load=format_wav.so             ; Microsoft WAV format (8000hz Signed Line
load=format_wav_gsm.so         ; Microsoft WAV format (Proprietary GSM)   
load=app_dial.so               ; Dialing Application                     
load=app_macro.so              ; Extension Macros                         
load=app_senddtmf.so           ; Send DTMF digits Application             
load=app_verbose.so            ; Send verbose output                     
load=pbx_config.so             ; Text Extension Configuration

BTW I noticed Google has a bug - placing a call to the Google Voice number resulted in my wife receiving an email "Peter Gunn wants to chat" (looks like this bug). Turns out that this is because I invited her to gmail and she is on my gmail chat list with "invited" status (you have to click the down arrow in chat last and select "show all contacts" to see this). The emails stopped once I blocked her in the list.

-PG

Thank you for the mini load list. Maybe I can move my Asterisk back to my DockStar based torrent box...

Thanks for the tutorials, petergunn and twinclouds. Asterisk installed without a hitch and with your downsized module set, it fits just fine in the Dockstar's 128MB memory. You two deserve gold stars or kudos or something.

FWIW, in a perfect world I would probably have put the Asterisk server on my SheevaPlug server, but I decided instead to use a spare Dockstar that I had laying around. Just completed a 50 minute Google Voice call through it and it worked very well indeed. I would presume that VoIP calls could be quite degraded if I had indeed used the SheevaPlug, because it also serves as a video recorder/server. Is my thinking correct here?

In any event, I'm delighted to be able to use Google Voice without having to be tethered to the computer. Thanks again!

FWIW, I did a bit of playing around with my new asterisk server today (thanks again, guys!). One problem I observed was that the incoming caller id was passed in the squirrelly Google format no phone understands, and thus it wasn't recognized by my phone as being from a known caller. I traced this down to the fact that several of the functions used to translate the Google format to the standard CID format in extensions.conf were missing. To fix this, I added the following modules to the minimal list petergunn gave out earlier:

load=func_callerid.so          ; function CALLERID()
load=func_db.so                ; function DB(), DB_EXIST()
load=func_math.so
load=func_channel.so
load=func_strings.so
load=func_logic.so

After doing this, the /var/log/asterisk/messages log was clean on both call origination and reception and caller ids were displayed correctly. (Of course, the name was not given, as I don't think Google pays for that lookup.) This change does make the asterisk server a bit bigger, though, but nowhere near what it was originally:

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND            
16222 root      20   0 16244 6768 4220 S  0.0  5.4   0:00.43 asterisk

FWIW. YMMV.

Oh, BTW, I also made a few (short) test calls today while streaming a 1080i video from the same Dockstar I was using for Asterisk, and the call was clean as far as I could tell. Jitter was minimal. I'm no Asterisk expert by any means, but someone I was chatting with about this said he thought Debian has some QoS code in the kernel and Asterisk makes use of it. (Does anyone know?) In any event, on first blush, it appears an Asterisk call is not affected nearly as much as I would have guessed by other traffic -- even fairly heavy traffic -- on the server's ethernet port.



Edited 1 time(s). Last edit at 03/07/2011 05:06PM by restamp.

From what I read asterisk can set the 802.1p QoS tags (part of the VLAN field) in the IP header to mark the audio packets as being prioritized. There is some config in iax.conf. I think iproute2 outgoing qdisc will ignore but preserve this when publishing the packet unless you add specific config (i.e. tbf or htb qdiscs).

Then its up to the network hardware. Surprisingly cheapo switches like the DGS-2208 do claim support for 802.1p QoS but I have no idea what that really means - they may just preserve the field when forwarding it.

Question: Do we really need apache to run asterisk?

-PG



Edited 1 time(s). Last edit at 03/07/2011 08:10PM by petergunn.

Thanks for the info on QoS, petergunn. I wondered the same thing about the need for Apache. Here's my best guess: twinclouds was running FreePBX along with Asterisk, and FreePBX would of course require Apache. I would be surprised if Asterisk needs Apache when running stand-alone, since it has no web-based interface.

Apache is not needed for plain asterisk, you are absolutly right.

Ok, I have removed apache and added restamps config to fix the CallerID. All working good :-)

Yesterday I upgraded my cell phone to an HTC EVO 4G (Android) and installed sipdroid so I can make and receive free wireless calls using my asterisk server. This works over 3G/4G/Wifi and does not use cell minutes. I installed the current version of sipdroid that was available through Android Market.

My dockstar is inside my home network which only has a dynamic IP address from my internet provider so the config is was a bit tricky:

You need 2 extensions in /etc/asterisk/sip.conf and you need to specify the externally visible IP address that sipdroid will use:

[general]
context=default                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
bindport=8024                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
externhost=petergunns_dynamic_host.dyndns.org ; dynamic hostname asterisk uses to find external IP
externrefresh=60                 ; refresh every 60secs in case dynamic IP changes
localnet=192.168.1.0/255.255.255.0 ; Class C LAN addesses
;tcpenable=yes                   ; Enable server for incoming TCP connections (default is no)
;tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls

[101]
; main home SIP phone connected to dockstar via Linksys PAP2
username=101
secret=passwordfor101
type=friend
callerid="Peter Gunn <101>"
host=dynamic
context=outbound
outgoinglimit=1
incominglimit=1
canreinvite=no

[102]
; PG's HTC EVO
username=102
secret=passwordfor102
type=friend
callerid="Peter Gunn <102>"
host=dynamic
context=outbound
outgoinglimit=1
incominglimit=1
nat=yes
canreinvite=no

Then modify /etc/asterisk/extensions.conf so it rings both the EVO and the home extension for incoming calls:

...
[google-in]
exten => peter.gunn@gmail.com, 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
exten => peter.gunn@gmail.com, n, NoOp(Callerid  ${CALLERID(name)})
exten => peter.gunn@gmail.com, n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten => peter.gunn@gmail.com, n, Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => peter.gunn@gmail.com, n, Dial(SIP/101&SIP/102, 180, D(:1))
exten => peter.gunn@gmail.com, n(bridged),Bridge(${DB_DELETE(gv_dialout/channel)}, p)
...

I also changed my SIP port (see sip.conf above) and RTP port ranges in /etc/asterisk/rtp.conf so I just needed to set up port forwarding for 8024-8050 on my home router:

[general]
rtpstart=8025
rtpend=8050

On sipdroid I set the SIP Proxy to petergunns_dynamic_host.dyndns.org port 8024. I had to disable PCMA audio codec in sipdroid config to make sipdroid outgoing calls work correctly.

-PG



Edited 4 time(s). Last edit at 03/12/2011 02:28PM by petergunn.

Also works good with the built-in native SIP client in Android 2.3 (gingerbread).

-PG

i was wondering if anyone could help me out. every time i try to dial out i get a error that says
[Mar 15 19:38:09] NOTICE[21056]: chan_sip.c:23357 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101
-- Executing ["the number"@outbound:1] Dial("SIP/101-00000000", "Gtalk/"my accout"/"the number"@voice.google.com") in new stack
[Mar 15 19:38:09] ERROR[21084]: chan_gtalk.c:1871 gtalk_request: No XMPP client to talk to, us (partial JID) : "my account"
[Mar 15 19:38:09] WARNING[21084]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'Gtalk' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/101-00000000' status is 'CHANUNAVAIL'

[Mar 15 19:38:09] NOTICE[21056]: chan_sip.c:23357 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101

^^ Seems harmless - just means there is no mailbox set up in /etc/asterisk/voicemail.conf

-- Executing ["the number"@outbound:1] Dial("SIP/101-00000000", "Gtalk/"my accout"/"the number"@voice.google.com") in new stack

^^ This is good.

[Mar 15 19:38:09] ERROR[21084]: chan_gtalk.c:1871 gtalk_request: No XMPP client to talk to, us (partial JID) : "my account"
[Mar 15 19:38:09] WARNING[21084]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'Gtalk' (cause 0 - Unknown)

^^ This is not good. I think something is wrong with your Jabber connection to Google Voice.

In /etc/asterisk/gtalk.conf you should have a "connection" setting that references the Jabber config:

[superm1-gtalk]
username=superm1@gmail.com
...
connection=superm1 ; this is the important bit that references the [superm1] section in jabber.conf

In /etc/asterisk/jabber.conf

[superm1] 
type=client
serverhost=talk.google.com
username=superm1@gmail.com/Talk ; this needs to be the right username
secret=superm1password ; this needs to be the right password for the google voice account
port=5222
...

Bear in mind I'm an asterisk n00b myself :-)

-PG



Edited 1 time(s). Last edit at 03/15/2011 10:17PM by petergunn.

Anthony, where are you seeing these messages? In /var/log/asterisk/messages?

I don't see any of them here, even the rather benign looking Notice. I rather suspect that you have one or more of your conf files in /etc/asterisk misconfigured. The ones that must be modified/personalized are: jabber.conf, sip.conf, gtalk.conf, and extensions.conf, plus modules.conf if you want to minimize the size of the asterisk daemon.

I would carefully recheck these files to verify that you've converted all the superm1 references to your own Google account, and not changed or deleted anything else in the process.

FWIW, my gtalk.conf is very simple:

[general]
context=google-in		; Context to dump call into
allowguest=yes

[guest]			; special account for options on guest account
disallow=all
allow=ulaw

I'm not saying it's correct, but this works for me.

Good luck with it. If you want to post your modified .conf files (after expunging your personal information), I'd be glad to take a look at them and see if I can spot anything suspicious.

On a slightly different topic, today, for security, I converted my Asterisk server to run under its own "asterisk" userid instead of "root". This may not be a comprehensive list of what needs to be changed to de-root a full-fledged Asterisk server, but it seems sufficient for what I am doing, namely using Asterisk solely as a conduit from my VoIP phones to Google Talk.

# adduser asterisk ...
# chgrp -R asterisk /etc/asterisk /usr/lib/asterisk /var/lib/asterisk
# chown -R asterisk:asterisk /var/log/asterisk
# chown -R asterisk:asterisk /var/run/asterisk
# chown -R asterisk:asterisk /var/spool/asterisk
# chown -R asterisk /usr/lib/asterisk/keys /usr/lib/asterisk/astdb    ## needed?
# chown -R asterisk /var/lib/asterisk/sqlite.db   ## needed? what is this?
# vi /etc/default/asterisk
	+ uncomment AST_USER and AST_GROUP

I then restarted the server:

/etc/init.d/asterisk stop
/etc/init.d/asterisk start

verified that the logs were clean, and made a couple inbound and outbound test calls. Everything seemed to work just like it did before.

Sorry I didn't come to this forum and thread lately so didn't answer some of the questions. For Asterisk, I tried make Openwrt working on Dockstar and with others' contributions, it works very reliably now and take a lot of resources. Moreover, no compilation is needed. However, this is outside of the scope of this forum.
I am glad what I did helped.

With the default extensions.conf file modifications provided by Mario Limonciello, 800 numbers do not go through. I'm not sure why. Mario uses ideasip.com to place them, and ideasip claims to pass SIP originated 800 calls for free and with no registration. In any event, I couldn't get ideasip.com to accept my calls, so I changed the [tollfree] section of the extensions.conf file from:

exten => _1800NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60) 
exten => _1888NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60) 
exten => _1877NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60) 
exten => _1866NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)

to:

exten => _1800NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _1888NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _1855NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _1877NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _1866NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})

denetron.com is another service that passes toll-free calls without cost and without requiring registration.

However, while playing around with this, I noticed quite by accident that Google also handles 800 numbers just fine. So, you can simply comment out the "include -> tollfree" entry under [outbound], and let Google route these calls.

Either way you can now make toll free calls; it's your choice. I chose Google, but if you prefer not passing your Google Voice number for some reason, denetron is the way to go.

And while you're at it, you might want to change the "512" area code in the [seven-digit] section to something more meaningful to you, although I suspect your ATA or SIP phone will have already done this in its dialplan.

Enjoy!



Edited 1 time(s). Last edit at 03/22/2011 12:20AM by restamp.

Peter,

Excellent noob work! you've helped me, a newer noob, so much with your posts.

Just a suggestion, in case possible, to further help some others who are putting together bits of info.

Your excellent guide here:
http://forum.doozan.com/read.php?2,1647,4063#msg-4063

Step #3, 2nd substep, seems to miss the necessary changes to gtalk.conf that you describe in the quoted material below.

Separately, I spent a couple hours reading through the google page on sipdroid about one way audio. The answer was here all along. Disabling that PCMA option worked for my xperia x10 mini and my g1. Time to throw my Linksys WIP330 into the electronics graveyard (my closet)!

Best Regards,
Jason


petergunn Wrote:
-------------------------------------------------------
> > NOTICE[21056]: chan_sip.c:23357
> handle_request_subscribe: Received SIP subscribe
> for peer without mailbox: 101
>
> ^^ Seems harmless - just means there is no mailbox
> set up in /etc/asterisk/voicemail.conf
>
> -- Executing ["the number"@outbound:1]
> Dial("SIP/101-00000000", "Gtalk/"my accout"/"the
> number"@voice.google.com") in new stack
>
> ^^ This is good.
>
> ERROR[21084]: chan_gtalk.c:1871 gtalk_request: No
> XMPP client to talk to, us (partial JID) : "my
> account"
> WARNING[21084]: app_dial.c:2039 dial_exec_full:
> Unable to create channel of type 'Gtalk' (cause 0
> - Unknown)
>
> ^^ This is not good. I think something is wrong
> with your Jabber connection to Google Voice.
>
> In /etc/asterisk/gtalk.conf you should have a
> "connection" setting that references the Jabber
> config:
>
>
> username=superm1@gmail.com
> ...
> connection=superm1 ; this is the important bit
> that references the section in jabber.conf
>
> In /etc/asterisk/jabber.conf
>
>
> type=client
> serverhost=talk.google.com
> username=superm1@gmail.com/Talk ; this needs to be
> the right username
> secret=superm1password ; this needs to be the
> right password for the google voice account
> port=5222
> ...
>
> Bear in mind I'm an asterisk n00b myself :-)
>
> -PG

Thanks Jason - I updated the guide so step 3) should replace superm1 with the right name.

-PG

Since everyone here uses the same hardware and OS, can someone share their compiled Asterisk binary (or better yet, a Debian package) for all to enjoy?

P.S. How stable is the setup? Can it be used as a practical landline alternative?

Ananon Wrote:
-------------------------------------------------------
> Since everyone here uses the same hardware and OS,
> can someone share their compiled Asterisk binary
> (or better yet, a Debian package) for all to
> enjoy?
>
> P.S. How stable is the setup? Can it be used as a
> practical landline alternative?

Yes. You can use .deb to install it with some additional file copying. Please see my latest post here: http://forums.plugpbx.org/index.php/topic,266.0.html. I have the .deb file generated and have wrote some scripts to automate the installation. However, I am not confident enough to publish them before further testing. If anyone is interested to help in testing it, I can send you a "beta" version if you pm me. Another issue is that it is about 26M in size. It is too large to send my email. Any suggestions? At this stage, I still want to have some control because it is not ready for public consumption yet.
As for the quality, I feel it is comparable to wireline in both connection delay and voice quality. Just don't want Google to change this free service in the future.

Twinclouds, email me and I can upload to http://Jeff.doozan.com/debian/lcd

-PG

petergunn:
Thanks. However, I would like to have some other to test it before public downloading. I will let you know once it is ready.
Please let me know your email address. You can pm me in the other forum if this forum does not have one.

Off-topic: twinclouds, as I was researching online more into the Asterisk on NAS devices I have also learned that we share a same hometown & employer in addition to our hobby :)

Interesting. Then please send me an internal email so we can keep in touch. (I assume you can find my email address, since you have already know so much about me.) If not I will tell you my first name next time. There are not many people have the same first name with me.



Edited 1 time(s). Last edit at 03/24/2011 11:59PM by twinclouds.

FYI - I tried the Android Market version of Linphone Video and it works as both a regular SIP extension and also does video calls to other SIP phones, but the sounds quality is awful on my HTC EVO 3D + the video comes from the back facing camera rather than the front facing one (changing the setting doesn't help).

To get video working you need to modify the asterisk config:

/etc/asterisk/modules.conf

load=format_h264.so

/etc/asterisk/sip.conf

nat=yes
videosupport=yes
allow=h264 ; only needed if disallow=all is set

Linphone defaults to MPEG4 codec but I didn't see a module under /usr/lib/asterisk/modules so I disabled it and selected H264 video. After I had uninstalled it I did notice that my asterisk does seem to have support for it ("core show codecs" in CLI) without specifying a specific module. Its possible this codec may have given better results.

Hopefully Linphone evolves into something usable soon :-)

-PG



Edited 2 time(s). Last edit at 03/27/2011 07:49PM by petergunn.

Peter:
I tried a few sip phones including the two you mentioned in your posts on my HTC Incredible. However, I always have problem to register with my Asterisk server. When you use a non-standard port (5060), how do you specify in the Linphone? Do you use serverip:port #? I Sipdroid, I always have the problem say no data network available even though I have both wifi and 3g service available.

In addition, I don't know if anyone can give me some more hint on how to register sip phone from internet. I can login using serverip:port # as the domain name and changed sip.conf accordingly. If I make a call, the other phone will ring but I cannot hear anything either way. Once I hear one way but not the other way. Both ways gone again.
Any guidance will be greatly appreciated.