TIPS incrediblePBX googlevoice trunks (no audio)

Timma

New Member
Joined
Jul 1, 2018
Messages
7
Reaction score
2
I hope i am posting in the right section

This week i am moving my asterisk server (v1.8 running on top of pfsense 2.3).
Up until last week this install was working with about five google voice accounts perfectly.
(Jabber/Gtalk modules)
It then decided one morning it would be totally unreliable, so i started looking at other solutions.
My home setup is pretty simple, a couple VLANs Trunked to PFsense. IPBX has a IP on the LAN designated interface. IncrediblePBX 13 on scientific linux was my choice, the install went fine. (pretty impressive so far)


I have setup the local extensions, calling local to local works fine.
I setup a google voice trunk using Oauth, it seems to connect fine.
made inbound and outbound rules, and as far as i can tell with the call logs and sip messages via asterisk (sip debug option) that calls are actually connecting. I don't see any warnings etc.

Here is the weird part, if i call a local PSTN number, the call connects but i have no audio in either direction.
but if i call my own GV number I can make calls fine via option 2. (audio in both directions.)

In my trouble shooting, I have tried setting the extension options to NAT. (no change.)
Setting the outside IP in the asterisk SIP section. (no change.)
trying with NAT settings on and off (no change.)
port-forwarding tcp/udp 10000 - 50000 to IPBX (no change)
port-forwarding tcp/udp 5222 to IPBX (no change)

Am i just missing something really simple?
Any advice would be greatly appreciated.
 

Waffull

New Member
Joined
Jul 2, 2018
Messages
29
Reaction score
1
Google is shitting down xmpp, that's why you're having audio problems. There is no fully vetted option yet, but you can give this a try: http://nerdvittles.com/?p=26204

PS: I have the above setup with mixed results on RasPi.
 

Seattle

New Member
Joined
Jul 4, 2018
Messages
3
Reaction score
0
I hope i am posting in the right section

This week i am moving my asterisk server (v1.8 running on top of pfsense 2.3).
Up until last week this install was working with about five google voice accounts perfectly.
(Jabber/Gtalk modules)
It then decided one morning it would be totally unreliable, so i started looking at other solutions.
My home setup is pretty simple, a couple VLANs Trunked to PFsense. IPBX has a IP on the LAN designated interface. IncrediblePBX 13 on scientific linux was my choice, the install went fine. (pretty impressive so far)


I have setup the local extensions, calling local to local works fine.
I setup a google voice trunk using Oauth, it seems to connect fine.
made inbound and outbound rules, and as far as i can tell with the call logs and sip messages via asterisk (sip debug option) that calls are actually connecting. I don't see any warnings etc.

Here is the weird part, if i call a local PSTN number, the call connects but i have no audio in either direction.
but if i call my own GV number I can make calls fine via option 2. (audio in both directions.)

In my trouble shooting, I have tried setting the extension options to NAT. (no change.)
Setting the outside IP in the asterisk SIP section. (no change.)
trying with NAT settings on and off (no change.)
port-forwarding tcp/udp 10000 - 50000 to IPBX (no change)
port-forwarding tcp/udp 5222 to IPBX (no change)

Am i just missing something really simple?
Any advice would be greatly appreciated.

I installed Asterisk 13 per http://nerdvittles.com/?p=26267 and there was audio problems when I used google voice on voip app, but extension to extension call worked fine. I played around the NAT setting and port forwarding and the problems were still there. Finally, I decided to give a shot on my Obi110 and the problems were gone. I do not understand why but it's working now.
 

chemcat9

Guru
Joined
Apr 19, 2010
Messages
111
Reaction score
4
I can only assume I'm impacted with the end of xmpp; my Incredible PBX 12.0.74 is in this same weird state where it will receive calls with two way audio, but when making calls it is intermittent with no outbound audio.

Will the http://nerdvittles.com/?p=26204 solution work with 12.0?

I'd like some feedback with this work around until a solution is working with backwards compatibility is confirmed. Is it possible to disable outbound via google and use a standard (DID4Sale, Vitelity, etc) call provider?

Much appreciated in advance - this forum has been an outstanding service to the VoIP community and I am much appreciative. Ward and the gurus' have done an awesome job keep us informed.
 

Laurence

New Member
Joined
Jul 6, 2018
Messages
17
Reaction score
0
I installed Asterisk 13 per http://nerdvittles.com/?p=26267 and there was audio problems when I used google voice on voip app, but extension to extension call worked fine. I played around the NAT setting and port forwarding and the problems were still there. Finally, I decided to give a shot on my Obi110 and the problems were gone. I do not understand why but it's working now.

Out of interest can you share you OBI110 settings? I am trying to use a Linksys SPA2102 and can't get it to connect to Asterix but the SIP soft client works okay
 

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
19,168
Reaction score
5,199
I can only assume I'm impacted with the end of xmpp; my Incredible PBX 12.0.74 is in this same weird state where it will receive calls with two way audio, but when making calls it is intermittent with no outbound audio.

Will the http://nerdvittles.com/?p=26204 solution work with 12.0?

I'd like some feedback with this work around until a solution is working with backwards compatibility is confirmed. Is it possible to disable outbound via google and use a standard (DID4Sale, Vitelity, etc) call provider?

Much appreciated in advance - this forum has been an outstanding service to the VoIP community and I am much appreciative. Ward and the gurus' have done an awesome job keep us informed.

Should work with Incredible PBX 12, but it hasn't been tested so make a backup. Yes, you can use a standard VoIP provider for outbound calling. That's the beauty of having a PBX. You can mix and match.
 

Timma

New Member
Joined
Jul 1, 2018
Messages
7
Reaction score
2
Thanks for the responses, sorry it took me so long to re-appear!

I wound up doing a fresh install of ipbx-13-13. (i wasn't very confident with all the settings i messed with in my frustration.)

after the clean install i followed the instructions from the link in the first response. (http://nerdvittles.com/?p=26204)

everything went fine, i wound up with the default 701 ext linked to my GV account. dialing 48 then a PSTN number works perfectly.(audio in both directions) in fact the call quality seems like its better than it has ever been.

inbound calling rings off the hook until GV-voicemail picks up.
so, no inbound whatsoever.

I have a computer graveyard at my disposal, and can put a good 10-15 hours a week towards testing/debugging. (assuming this isn't a problem with my network configuration.) let me know, i don't mind if it helps the user community as a whole.

I have seen mention of google-voice sip gateway services. are they now required to get this working?

if anyone can think of anything else to try and troubleshoot this, don't hesitate to reach out.

Thank you in advance for any leads to get this working again!
 

Adam Ringer

New Member
Joined
Aug 27, 2015
Messages
2
Reaction score
0
I am also noticing this, GV is setup with OAuth on 13-3 install from ISO. Inbound calls seem to be ok with bidirectional audio, but on outbound calls there is no audio from the remote phone to the PBX on numerous GV numbers I have tried.
 

Waffull

New Member
Joined
Jul 2, 2018
Messages
29
Reaction score
1
I have seen mention of google-voice sip gateway services. are they now required to get this working?
/QUOTE]

The gateway you've seen mentioned is simonics GVGW. Unfortunately he has/is shutting that down due to issues migrating all his users from his xmpp gateway to a sip gateway. The good news is with the development by naf of his patch, Asterisk can connect direct, so no gateway is needed.

As for your incoming call problem, the only time I've experienced that is when I attempt to setup multiple IP phones/devices with the same extension. In that case, the last device to register with Asterisk will be the only one receiving calls. Prior to my switching to incrediblePBX and naf's patch, I was able to utilize multiple devices per extension by using chan_pjsip extensions and setting Max Contacts to > 1. That worked on RASPBX, but isn't working with IncrediblePBX for some reason.

As for your network, unless you are specifically blocking or forwarding 5060/5061, NAT will be utilized for the transport tunnels. I'm anything but a VoIP/Asterisk expert... far from it. So beyond this I can't offer much help.. but if you haven't, you might want to read through the gvsip thread over on dslreports for any clues to what may be the issue: https://www.dslreports.com/forum/r3...le-Voice-SIP-testing-and-technical-discussion
 

Adam Ringer

New Member
Joined
Aug 27, 2015
Messages
2
Reaction score
0
I am also noticing this, GV is setup with OAuth on 13-13 install from ISO. Inbound calls seem to be ok with bidirectional audio, but on outbound calls there is no audio from the remote phone to the PBX on numerous GV numbers I have tried.

I hastily replied earlier, I believe I need to configure the GV SIP, but as I have more than 2 trunks, I will patiently await the ability to set that up

EDIT: Everything is ok now, was able to get all my GV accounts working with pjsip and a lot of config file editing.
 
Last edited:

Timma

New Member
Joined
Jul 1, 2018
Messages
7
Reaction score
2
I hastily replied earlier, I believe I need to configure the GV SIP, but as I have more than 2 trunks, I will patiently await the ability to set that up

EDIT: Everything is ok now, was able to get all my GV accounts working with pjsip and a lot of config file editing.

did you have to do anything out of the ordinary to get both incoming and outgoing working? (other than a non-gui funtastic voyage.)

i am quite familiar with editing the configs, as that was the only way to set it up on the last platform i was using.

even posting working config files may be a huge help to me, and anyone else searching this forum to solve this problem.

TIA~!
 

Timma

New Member
Joined
Jul 1, 2018
Messages
7
Reaction score
2
:sorcerer:very happy to report that it's working. Inbound & outbound audio in both directions.

I will add some other extensions this weekend. And try with two gv accounts.

For anyone else reading:

Ipbx13-13 fresh install.

Follow the link ward posted, or wait for the new gvsip-naf installer

Follow the script prompts, insert refresh key & GV number.

Make a new test extension. I'm using pjsip. Make inbound and outbound rules linking to the gvsip trunk.

Make some test calls...
Mine worked immediately, no NAT fiddling, no messing with config files.

Thank you very much @wardmundy
 

Timma

New Member
Joined
Jul 1, 2018
Messages
7
Reaction score
2
A big thank you to all involved with this!

I am back up and running with more features than ever!

Better call quality then I have ever had with my phone system.

I can't wait to tinker with some of the pbx features I have seen so far. Lol apparently a whole lot has happened since asterisk v1.8

@wardmundy, thanks again for all of your efforts!
 

Members online

No members online now.

Forum statistics

Threads
25,779
Messages
167,505
Members
19,199
Latest member
leocipriano
Get 3CX - Absolutely Free!

Link up your team and customers Phone System Live Chat Video Conferencing

Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.

3CX
A 3CX Account with that email already exists. You will be redirected to the Customer Portal to sign in or reset your password if you've forgotten it.
Top