Minimum implementation of Asterisk for Google Voice?

Is it possible port 5060 still forward to DockStar and not the router?

Not likely. You really don't need to forward any port for the basic setup to work.

Thanks for the help! I finally figured out - spurred by the posts here. The problem ended up being the directory references in asterisk.conf - since I installed asterisk on my router (optware), the directories needed to reference /opt/ rather than / - so the references in asterisk.conf neede to reference, for example, /opt/etc/asterisk rather than /etc/asterisk/

I knew it was something simple!

Thanks for the help - all is working now!

I am glad to hear everything works now. I guess I should put a note on my original post. Did you used my previous posts? If you did, which one?

For those using asterisk to make GV calls, there is an issue again. GV changed something and it does not work again without the patch mentioned here in Page 3. The best solution is from this blog with a patch to handle both situations.

https://michigantelephone.wordpress.com/2011/08/19/if-your-asterisk-server-stopped-putting-through-outgoing-google-voice-calls-today-here-is-the-fix-%E2%80%94-maybe/



Edited 1 time(s). Last edit at 10/13/2011 11:15AM by davidnewton.

Thanks, davidnewton. This seems to have done the trick for now and gotten my Asterisk happily connecting with Google Voice once again, but I wonder if Asterisk is going to eventually add some additional code to their GV to make future such changes bulletproof.



Edited 2 time(s). Last edit at 10/13/2011 05:58PM by restamp.

Hi davidnewton and restamp
If possible, please upload your working chan_gtalk.so. I have error to compile it which takes time to fix. Your working chan_gtalk.so will give rest of us a quick fix by downloading it.

Thank you very much.

dchen: This is the chan_gtalk.so file for 1.8.7.0. I am not sure if it works for other versions. Good luck!



Edited 1 time(s). Last edit at 10/14/2011 08:23AM by davidnewton.

Hi davidnewton

Thank you very much.
I tried it in my asterisk 1.8.6.0 which does not work.
If anyone has a working chan_gtalk.so for 1.8.6.0 or 1.8.5.0, please upload it to help me. I have error to compile it which takes time to fix.

Thank you very much.

dchen4649 Wrote:
-------------------------------------------------------
> Hi davidnewton
>
> Thank you very much.
> I tried it in my asterisk 1.8.6.0 which does not
> work.
> If anyone has a working chan_gtalk.so for 1.8.6.0
> or 1.8.5.0, please upload it to help me. I have
> error to compile it which takes time to fix.
>
> Thank you very much.

dchen: I am trying to find a place to upload the 1.8.7.0 deb file for you to install with dpkg. It is about 9MB (without sounds package). If you know any free places allowing 9mb attachment, please let me know.

UPDATE: BTW, I found out that dslreport forum can upload my deb file. Please go to this site to download and try it.

http://www.dslreports.com/forum/r26425026-Asterisk-GV-call-in-asterisk-does-not-work-again



Edited 3 time(s). Last edit at 10/14/2011 10:06AM by davidnewton.

Hi davidnewton,

You are great.

Thank you very much and have a nice weekend.

davidnewton Wrote:
-------------------------------------------------------
> dchen4649 Wrote:
> --------------------------------------------------
> -----
> > Hi davidnewton
> >
> > Thank you very much.
> > I tried it in my asterisk 1.8.6.0 which does
> not
> > work.
> > If anyone has a working chan_gtalk.so for
> 1.8.6.0
> > or 1.8.5.0, please upload it to help me. I have
> > error to compile it which takes time to fix.
> >
> > Thank you very much.
>
> dchen: I am trying to find a place to upload the
> 1.8.7.0 deb file for you to install with dpkg. It
> is about 9MB (without sounds package). If you know
> any free places allowing 9mb attachment, please
> let me know.
>
> UPDATE: BTW, I found out that dslreport forum can
> upload my deb file. Please go to this site to
> download and try it.
>
> http://www.dslreports.com/forum/r26425026-Asterisk
> -GV-call-in-asterisk-does-not-work-again
Your deb file work very well on my dockstart. Is there any trick in building the package? I was able to build asterisk 1.8.2.3 a while back. For some reason, the binaries I "made" for 1.8.7.0 or 1.8.7.1 don't work. When a softphone tries to register to asterisk, it doesn't show any activities. I can get to asterisk's console, just don't see any activites.

funtoy1001 Wrote:

> Your deb file work very well on my dockstart. Is
> there any trick in building the package? I was
> able to build asterisk 1.8.2.3 a while back. For
> some reason, the binaries I "made" for 1.8.7.0 or
> 1.8.7.1 don't work. When a softphone tries to
> register to asterisk, it doesn't show any
> activities. I can get to asterisk's console, just
> don't see any activites.

There is no tricks for me. I just followed twinclouds' instruction here to compile it. After downloading the new package, three files need to be modified (main/utils.c, makeopts.in as pointed out by twinclouds, and the new patch for channels/chan_gtalk.c).

http://forums.plugpbx.org/index.php/topic,247.msg1168.html

It appears that Google change the gtalk setting to the original one again(!). Fortunately, in installed (from China :-)) the 1.8.7.0.deb file provide by davidnewton and it works. Thanks! Hopefully I don't have to modify it again.
Not sure what game Google is playing.

Just installed Asterisk 1.8.8.1 following the instruction of TC and PG. Everything works great except:
1. cannot make any tollfree call and get the this message:

*CLI>   == Using SIP RTP CoS mark 5
    -- Executing [8004267378@outbound:1] Set("SIP/101-00000000", "CALLERID(dnid)=18004267378") in new stack
    -- Executing [8004267378@outbound:2] Goto("SIP/101-00000000", "18004267378,1") in new stack
    -- Goto (outbound,18004267378,1)
    -- Executing [18004267378@outbound:1] Dial("SIP/101-00000000", "SIP/18004267378@proxy.ideasip.com,60") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/18004267378@proxy.ideasip.com
[Jan  6 07:17:01] WARNING[1001]: chan_sip.c:3629 retrans_pkt: Retransmission timeout reached on transmission 7b3b88ff7b9f362d4f1dbf0c2261b926@192.168.2.20:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jan  6 07:17:01] WARNING[1001]: chan_sip.c:3658 retrans_pkt: Hanging up call 7b3b88ff7b9f362d4f1dbf0c2261b926@192.168.2.20:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/101-00000000' status is 'CHANUNAVAIL'

2. significant delay (5-15 secs) from the dialing time to any response from the Asterisk server.

Anyone experiencing the same thing? Or does anyone know how to fix this?

jrey - i used it y'day and i didnt have any problems.

is there any commands that i could use to check status on clients registered or calls in progress or call statistics etc.. on asterisk?

exenew, the client is registered ok and I can call any non-tollfree number. I think something is wrong with the ideasip.com for the tollfree. Are you using ideasip for those calls?

Very nice Article!

Google Talk and Asterisk 1.8.7.1
hope it helps!!

http://www.opentut.com/?p=1#comments

Dear All,

Looking for help again and hoping someone may have seen similar.

Background:
Installed asterisk pretty much exactly like petergunn
asterisk is installed on a dockstar
Have two lines set up. One 'household' phone and one on an android cellphone


The problem lies with the Android cellphone. Like many of the instructions here I have set up a tunnel in my router, changed ports, and configured the codecs. Additionally, I added the line #2 to ring out in the appropriate config file.

The problem:
When I am on my work WiFi with my cellphone, I am unable to receive a call to my google voice phone number. However, I am easily able to place a call from it. Equally starnge is that when my cellphone is on my home wifi, on same subnet as my asterisk server then I am able to both place and receive calls.

On the Android phone I am running sipdroid. In both cases, home and work, the phone registers well with the asterisk server and the Android status shows 'Preferred'. I looked through the log and I see some message about unable to find the client or something, but no reason is given. Does anyone have any experience like this? Anyone can recommend some things to look into at my router, at the Android phone, or at the Asterisk server? The only point of failure I won't be able to look into in detail will be the work firewall/router/settings/etc.

Today I will try it at starbucks and check. Thank you all in advance for advice or pointers you can provide. It'd be great for me to be able to receive incoming calls over gtalk while I am in China in a couple days.

I'm not expert but if its ringing when you get an incoming call but you cant answer it then your work firewall is filtering the UDP voice traffic (RTP) for incoming calls. With Asterisk the signalling (SIP) can go over TCP or UDP but voice always goes over UDP and firewalls will not normally allow incoming UDP unless there is a static route or there was a recent outgoing TCP or UDP connection from inside the firewall from your NATted IP.

If you dont even get notified of incoming calls then the firewall is probably blocking incoming SIP UDP as well. For SIP you could reconfigure asterisk and sipdroid to use TCP/IP but that wont help with the RTP voice traffic which is always UDP. Worth trying though if you dont get notified of incoming calls.

If the firewall is the problem and simple tweaks (move the RTP port range around, use TCP for SIP, etc.) dont work then you're only option might be to connect over a VPN connection (i.e. openVPN) to your asterisk server. This will tunnel UDP and/or TCP back to your home network where asterisk is running. I have openVPN running on my dockstar and I believe there is an android openVPN client (I may have tried it - cant remember) but I have never tried sipdroid over openVPN on Android. In theory it might work although the latency might make it unuseable. On the plus side you wouldnt have to worry about people eavesdropping :-)

If you were mostly stationary you could run a VPN endpoint and hostapd on a laptop and connect to that via sipdroid over wifi, or maybe dd-wrt flashed a wireless access point set to repeater mode. Just brainstorming - the Asterisk forums would probably have more practical things to try.

-PG

I have just installed Asterisk with a Cisco SPA112 ATA following twincloud and Peter Gunn's comprehensive guides.

It's running in Ubuntu 10.04 (32bit) which is as a guest hosted using VMware player under Ubuntu 12.04 (64bit).

Thank you both very much indeed!

(I originally tried installing the AsteriskNOW iso but Centos wasn't cooperating - the xserver was causing problems preventing the machine from rebooting after the install, so I gave up on that route).



Edited 1 time(s). Last edit at 06/15/2012 03:35PM by simonrodan.

Alrighty. I have been trying to get this setup for several days. I have read many different tutorials and I think I have a decent understanding of what I need to configure.

My real problem is I get no incoming calls from my google voice number to my phones. Outgoing calls work fine. Calls between extensions work fine(although this technically isn't google voice, it leads me to believe the various phones are setup correctly.

I believe I have follow several tutorials very closely and I just cannot get that to work.

I have been watching the debug info (started a monitor with 'asterisk -rvvvvv -ddddd' and put in sip set debug on. There was one time (yesterday) when it reported something was trying to connect to it.

I am currently using asterisk 1.8.10.1 This was the last version before the asterisk folks declared google too much of a pain to keep up with and were going to leave it up to the community to support that, I did try the latest version (1.8.13.0) but it no longer seems to have the gtalk channel stuff included, at least it doesnt show up in menuconfig before you compile.

Is this something that isn't working? If it is working and I am just failing anyone who could push me in the right direction I would be grateful.

Thanks.

It is well known that gtalk incoming call is not very reliable for Asterisk. People suggest to use FreeSwitch but I never try it. I use gtalk for out calling anyway.
I installed Asterisk 13.0 recently and everything works fine. Have you installed libiksemel-dev and included chan_gtalk in the modules.conf?

I do have libiksemel-dev installled

root@Goflex-80G:/var/log/asterisk# dpkg --get-selections | grep libik
libiksemel-dev                                  install
libiksemel3                                     install

chan_gtalk is not explicitly stated in my modules.conf

[modules]
autoload=yes
noload => pbx_gtkconsole.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_console.so

I put this back to the default when changing it to the shortened version had it not working. I figured best to leave it as is until it works then changing it later.

When you say everything is working fine in 13.0 , do you also mean incoming calls? Do you still have to set PROC=arm? I needed to do that in the version I just installed.

Is there a Freeswitch for dockstar/goflex devices? I currently have no home phone and just want to use a single number that will call my cell when out and ring a phone in my house when home.

optim Wrote:
-------------------------------------------------------
> I currently have no home phone and just want to
> use a single number that will call my cell when
> out and ring a phone in my house when home.

Thats exactly what my set up does (rings home phones via LinksysPAP2 and my Android cell via Sipdroid), so it does work. I'm still using asterisk 1.8.3 with the outgoing calls patch.

-PG

One other question, I have seen both ways. The username (in jabber.conf?) should it have /talk at the end of it or not?

I forget off the top of my head, but there is a place where the instructions we have here do not have the /talk at the end of the gmail.com username but everywhere else that I saw it, it had .talk at the end.

optim Wrote:
-------------------------------------------------------
> One other question, I have seen both ways. The
> username (in jabber.conf?) should it have /talk at
> the end of it or not?

My jabber.conf looks like this:

[general]

[peter.gunn]
type=client
serverhost=talk.google.com
username=peter.gunn@gmail.com/Talk
secret=petergunns_password
port=5222
usetls=yes
usesasl=yes
statusmessage=PG-ast
status=xaway

[someone.else]
type=client
serverhost=talk.google.com
username=someone.else@gmail.com/Talk
secret=some_other_password
port=5222
usetls=yes
usesasl=yes
statusmessage=SE-ast
status=xaway

-PG

ok, I have added /Talk to the end of mine. Here is my gtalk.conf:

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

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

[rottstein-gtalk]
username=rottstein@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=rottstein

Does that look right?

optim Wrote:
-------------------------------------------------------
> ok, I have added /Talk to the end of mine. Here
> is my gtalk.conf:

> [rottstein-gtalk]
> username=rottstein@gmail.com
> disallow=all
> allow=ulaw
> context=google-in # PG: I dont have this, but it shouldnt matter
> connection=rottstein

> Does that look right?

Yes, looks good to me.

-PG



Edited 1 time(s). Last edit at 06/24/2012 10:19AM by petergunn.

I don't use it for calling in so I don't know if it works for 1.8.13.0 or not.
For 1.8.13.0, you don't need to do the PROC setting any more.
You can check if gtalk is loaded or not by doing "module show" in the CLI prompt.