[GUIDE] Setup Your Own Asterisk Server With Google Voice on Amazon EC2

[ryuker]

I posted this in the PIAF thread since I used that method but this will help you guys as well.


Nexus 4 SIP echo fix - It's pretty bad for many users but this solved it 100% for me



It worked for me and I tested numerous different setting and configurations along with measuring my latency to ensure it works. YMMV but it's easy to try. At first I thought it was a Latency issue since echo didn't occur over wifi but with ping times lower than 60ms I still got echo over 3g without this method.

I'm using the silk codec(Silk 24 wifi/Silk 12 or 16 over 3g/4g); YMMV with other codecs.



*Before starting*

Root your phone, flash radio .27 or .33 if your on T-mobile or in Canada: We want to enable Nexus 4g! 3g is usually fine but 4g tends to have significantly lower latency(3G: 50-70ms versus 4g: 20-40ms). Sometimes my 3g speed was fine but latency sucks(100-250+ms)....3g has enough speed for VoIP but any latency issues/spikes/jumps will ruin a call.

This is a good guide to follow: Hippowise"s guide


For the first few steps plus a audio fix some users encountered with the radio downgrade; there are 2 scripts you can run via any terminal emulator app scripter(I used this $Scripter). The actual Script to be run can be found on XDA in the nexus 4 LTE thread(download or save them as txt files). The XML file I suggest editing your self(I just replaced all instances of epc.t-mobile with fast.t-mobile for T-mo users).

If said "scripting...wut?" just follow Hippowise's guide to the tee. Scripting is just faster/easier.

It took a few days for my 4g/LTE to kick in so be patient; It's worth the trouble!


**Now to the guide**

First, get the Faux kernel for Nexus 4 or use another kernel that has applied his patch.

2nd, get his apps from the Play store(they cost money but worth it): Faux Speakers and Faux Sound: Configure them for "voice" and "quality" respectively(you can adjust on your own from there, I upped the gain a bit).

3rd, Use Csipsimple and enable "Software volume control" instead of the default android/os/hardware based volume/gain control. I wonder if this bypasses Faux and puts volume control solely in the hands of Csipsimple. I would do 1-3 steps anyway because it fixes the SQ issues in general with the nexus 4.

4th. Enable Echo cancellation; until I did the previous steps, echo cancelation didn't work at all: YMMV.

I choose Echo mode Webrtc becuase Speex echo cancellation didn't work for me. If your codecs are different the other choices or auto might work better. Experiement.

5th: *Optional*

Noise Cancelation: Worked so-so....didn't need so I turned off
Voice Detection: I didn't like it but experiement.

Compact SIP packets: I use it, optional
STUN: I turned it on, I don't think it will hurt.

Audio troubleshooting in the Csip menu has tons of options to mess with. If all else fails, read some guides and experiment. I didn't need to chance anything here but some of it looks promising and has some fixes for specific models like the galaxy.

6TH: ??????

7th: Latency

Get Speedtest app from the play store and test your speed. In some areas over 3g I would get decent down/up speeds but godawful latency (300ms+ ). If your 3g latency consistently sucks, do the 4g fix for the nexus; Much better latency and I always get 4g in most places on T-mobile.....rural areas might be screwed on getting 4g although I noticed my 3g speeds increased overall after the fix.

8th

Finished; report back in the thread and tell me if it helped!! And if anybody knows a good I.T. position open for a VoIP office/call center; let me know cause I'm quitting mine.

 

[ftherese]

Thanks a lot for the config - "one" lingering issue for me

extensions.conf

[general]
autofallthrough=yes

; If an unauthenticated request some how gets through, send them to free 411.
[default]
exten => 411,1,Answer()
same => n,Dial(Motif/google/1800...@voice.google.com)

[local]
exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)
exten => _XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)
exten => _+1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)

[incoming-motif]
exten => s,1,NoOp()
 same => n,Set(crazygooglecid=${CALLERID(name)})
 same => n,Set(stripcrazysuffix=${CUT(crazygooglecid,@,1)})
 same => n,Set(CALLERID(all)=${stripcrazysuffix})
 same => n,Dial(SIP/USERNAME,20,D(:1))

I am using this set up on my home asterisk setup, connecting with my android phone running sipdroid. Calling out works just fine, but when incoming calls ring my sipdroid, and I answer them I get "no data," while the calling phone just continues ringing until it gets to Google Voice-mail. I finally, today, had the idea of dialing "1" on my sipdroid after answering the call (even though there is no voice prompt or anything) and, voila it works. I thought that was strange because the Dial command under incoming-motif adds a D1), and that comes across in the log as well:

 -- Executing [s@incoming-motif:1] NoOp("Motif/+XXXXXXXXXXXX", "") in new stack
    -- Executing [s@incoming-motif:2] Set("Motif/+XXXXXXXXXX", "crazygooglecid=+XXXXXXXXX@voice.google.com/srvenc-92PWyN+7+//TmnbqAeG1ZyZUOAkykV8i") in new stack
    -- Executing [s@incoming-motif:3] Set("Motif/+XXXXXXXXXX", "stripcrazysuffix=+XXXXXXXXXX") in new stack
    -- Executing [s@incoming-motif:4] Set("Motif/+XXXXXXXXX", "CALLERID(all)=+XXXXXXXXX") in new stack
    -- Executing [s@incoming-motif:5] Dial("Motif/+XXXXXXXXXX", "SIP/ftherese,20,D(:1)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ftherese
    -- SIP/ftherese-00000000 is ringing
    -- SIP/ftherese-00000000 is making progress passing it to Motif/+XXXXXXXXXXX
[Sep 14 22:43:00] NOTICE[19355][C-00000000]: chan_motif.c:1636 jingle_indicate: Don't know how to indicate condition '14'
    -- SIP/ftherese-00000000 answered Motif/+XXXXXXXXXXX
    -- Sending DTMF '1' to the calling party.
    -- Locally bridging Motif/+XXXXXXXXXXXX and SIP/ftherese-00000000
  == Spawn extension (incoming-motif, s, 5) exited non-zero on 'Motif/+XXXXXXXXXXX'

Why do I have to dial "1" in my sipdroid to connect the call?

 

[omegatotal]

So has anyone had any luck figuring out the google voice motif timeout issue where it will not detect the connection status of the chan_motif module??

I am leaning towards just scripting an automated test call to something like 909-390-0003 (loopback audio test?) and if it fails to connect reload the chan-motif module (I can confirm that this call is passed and auto answered

   -- Called Motif/gmailtunnel/19093900003@voice.google.com
    -- Motif/19093900003@voice.google.com-4f30 is proceeding passing it to SIP/5003-00000001
    -- Motif/19093900003@voice.google.com-4f30 answered SIP/5003-00000001

but I suck at scripting :/

---------- Post added at 08:36 PM ---------- Previous post was at 08:03 PM ----------

Also a note on the costs

If you are running a regular m1-micro instance without reserving up front, thats $14.64/mo unless you qualify for the free tier which gives you a discount of $-16.84/mo

EBS Volumes, elastic IP's and iop's are extra but trivial, like a few dollars a month for 20GB and 10 iops/sec average over the month

data transfer is stupid cheap too like pennies per gb

if you reserve upfront the m1.micro for 1 year, costs goes down to an effective monthly cost of like 10.71 however you get next to no free tier discount if you choose this and there is a $23 upfront setup fee for 1 year,

REF:
http://calculator.s3.amazonaws.com/calc5.html

 

[Nick123194]

What do you do with the pem file?

Sent from my SPH-L720 using XDA Premium 4 mobile app

 

[davidnewton]

I got the same problem as this. Any solution?

I am using this set up on my home asterisk setup, connecting with my android phone running sipdroid. Calling out works just fine, but when incoming calls ring my sipdroid, and I answer them I get "no data," while the calling phone just continues ringing until it gets to Google Voice-mail. I finally, today, had the idea of dialing "1" on my sipdroid after answering the call (even though there is no voice prompt or anything) and, voila it works. I thought that was strange because the Dial command under incoming-motif adds a D1), and that comes across in the log as well:

 -- Executing [s@incoming-motif:1] NoOp("Motif/+XXXXXXXXXXXX", "") in new stack
    -- Executing [s@incoming-motif:2] Set("Motif/+XXXXXXXXXX", "crazygooglecid=+XXXXXXXXX@voice.google.com/srvenc-92PWyN+7+//TmnbqAeG1ZyZUOAkykV8i") in new stack
    -- Executing [s@incoming-motif:3] Set("Motif/+XXXXXXXXXX", "stripcrazysuffix=+XXXXXXXXXX") in new stack
    -- Executing [s@incoming-motif:4] Set("Motif/+XXXXXXXXX", "CALLERID(all)=+XXXXXXXXX") in new stack
    -- Executing [s@incoming-motif:5] Dial("Motif/+XXXXXXXXXX", "SIP/ftherese,20,D(:1)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ftherese
    -- SIP/ftherese-00000000 is ringing
    -- SIP/ftherese-00000000 is making progress passing it to Motif/+XXXXXXXXXXX
[Sep 14 22:43:00] NOTICE[19355][C-00000000]: chan_motif.c:1636 jingle_indicate: Don't know how to indicate condition '14'
    -- SIP/ftherese-00000000 answered Motif/+XXXXXXXXXXX
    -- Sending DTMF '1' to the calling party.
    -- Locally bridging Motif/+XXXXXXXXXXXX and SIP/ftherese-00000000
  == Spawn extension (incoming-motif, s, 5) exited non-zero on 'Motif/+XXXXXXXXXXX'

Why do I have to dial "1" in my sipdroid to connect the call?

I have exactly the same problem. If I use CSipSimple or Linphone, then there is no such problem. Probably SipDroid has some issues?

 

[jauling]

howdy all.

With the impending XMPP shutdown on May 15h, does this mean this thread basically is kaput? Are there any alternate methods to get SIP working post-shutdown?

 

[bluespire]

howdy all.

With the impending XMPP shutdown on May 15h, does this mean this thread basically is kaput? Are there any alternate methods to get SIP working post-shutdown?

SIP with GV? Probably not. I had good luck with Anveo. They are really cheap. US Phone # + e911 was about $5/month IIRC. I'll be moving back to that when GV quits. They offer Fax and Texting, as well, but last I checked their texting service was provided by a UK company, so no US number for that. If that doesn't both you then I would forge onward. There a number of other providers of course, my preference was Anveo.

I think that it's quite possible that Google will unveil a pay service to keep XMPP enabled on a per account basis, or maybe it will still be available with a business account. I guess we will have to see.

 

[bananaboat23]

Is this still possible to do with the end of Google's XMPP support?

 

[acegolfer]

Is this still possible to do with the end of Google's XMPP support?

I don't use this method any more. But dslreport ppl say that Google XMPP still works in some cases, and I think this method should work.

 

[tech128]

I don't use this method any more. But dslreport ppl say that Google XMPP still works in some cases, and I think this method should work.

It works in all cases. If it doesn't work for you, that means you need to troubleshoot.

It's a setting on google's server and/or your server.

Granted the new Asterisk 11 & 12 seem to be very retarded with respects to GV.

I was able to dial out no problem with it, but calls in wouldn't even get to the asterisk server for some reason. Very odd.

And don't get me started on this "motif" stuff they have now. Very confusing. gtalk and jabber were soooooooooooo much better/easier.

No such issues with 1.8.x

 

[omegatotal]

The problem with the current setup ( I use it ) is that if you sign into the email account it has a high probability of changing the destination for the incoming calls to the email client and the motif plugin for freepbx does not have a configuration option for timeouts

the best thing you can do is just script an rcon to force a reload if there are no active calls every night (which I have been to lazy to do)

It works in all cases. If it doesn't work for you, that means you need to troubleshoot.

It's a setting on google's server and/or your server.

Granted the new Asterisk 11 & 12 seem to be very retarded with respects to GV.

I was able to dial out no problem with it, but calls in wouldn't even get to the asterisk server for some reason. Very odd.

And don't get me started on this "motif" stuff they have now. Very confusing. gtalk and jabber were soooooooooooo much better/easier.

No such issues with 1.8.x

 

[tech128]

The problem with the current setup ( I use it ) is that if you sign into the email account it has a high probability of changing the destination for the incoming calls to the email client and the motif plugin for freepbx does not have a configuration option for timeouts

the best thing you can do is just script an rcon to force a reload if there are no active calls every night (which I have been to lazy to do)

Well if you're having issues with the email client overriding the incoming calls then that's a problem with the priority setting for motif.

It probably sets it below the priority for the email client.

The simonics gateway ran into that problem as well and managed to change the priority higher than the email client's.

Here is the relevant message: https://simonics.com/2012/11/gv-gateway-registration-open-new-features/

So if you can set the priority with motif or whatever they're using atm, then you should be set.

 

[tarun1188]

Unable to install asterisk

Hi, I followed all the steps but when i try to stop/start the asterisk server it says:

Stopping Asterisk PBX: asterisk
No /usr/sbin/asterisk found running; none killed. [ OK ]​

* Starting Asterisk PBX: asterisk Privilege escalation protection disabled!
Illegal instruction​

Also the contents of /var/run/asterisk is empty.

Can anyone help me setting this up. Let me know if i have to provide any more information.

Thanks in advance