TIPS Wazo+GV = No audio

[rogerjbos]

I am using Wazo 17.16 installed on a dedicated PC using the latest ISO. I set up a google voice trunk and can make and receive calls, but I don't hear any audio on their side. My Zoiper softphone shows I connect using uLaw and I am sending bits, but my recieving bits stays at zero. Any help would be appreciated.

 

[wardmundy]

How did you set up the GV trunk? Are you using Simonics?? Wazo 17.16 no longer supports Google Voice directly. You have to use a SIP gateway.

 

[rogerjbos]

OK. No, I am not using Simonics. I am surprised it even rings then. I thought it was a codec issue. Thank you.

 

[wardmundy]

Tell us how you set it up and we will experiment a little. Thanks.

 

[rogerjbos]

I just signed up with Simonics and I will test it later, but I am sure it will work with Simonics.

 

[Sylvain Boily]

Hello, normally with wazo 17.16 you can use directly the chan_motif to use google voice, we are using the asterisk 15.1 which embeded the google oauth2 directly without any patch. I haven't did test, but if someone have an account and the latest wazo version, i can check with him. In wazo 17.15 it's not possible to do that.

 

[rogerjbos]

If it is possible to use GV with wazo 17.16, I am willing to keep trying to get it to work. I have pasted the "message" that I am currently getting. I don't have any thing configured under SIP trunks. Under Income Calls I have my GV number in the DID field and under Outgoing Calls I have Preprocess subroutine: subr-gv-outcall and one trunk selected: motif / norfieldpbx | @voice.google.com.


[Nov 29 20:25:44] WARNING[16594] chan_sip.c: Retransmission timeout reached on transmission qjLgFKPBvQ9YAlONgx2i3w.. for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Nov 29 20:25:44] WARNING[16594] chan_sip.c: Hanging up call qjLgFKPBvQ9YAlONgx2i3w.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

 

[rogerjbos]

I also tried to configure my Simonics gateway using this guide: http://nerdvittles.com/?page_id=18280 and I get the same result. I call my work and hear the number ringing, when the voicemail picks up I don't hear any audio. Packets are being sent, but no packets are being received.

 

[billsimon]

It sounds like you haven't configured the NAT settings on your PBX. One-way or no-audio conditions are usually a symptom.

 

[Eliad]

When I was using Wazo I had similar issues. I suppose your Wazo PBX is behind a firewall.
I have PFSense firewall. I had to do do a port forward to the Wazo server for port 5060.
I had also an issue with Simonics. At random the calls would go to Google voice instead of being picked up by the Wazo. Then I used OAuth for the google voice and with this one I never got directed to the google voice unless Wazo server was down.

 

[Eliad]

To be more exact my firewall has UDP port forward enabled for 5060-5069 and 10000-50000. Hope this helps.

 

[rogerjbos]

Clearly there is something wrong with my setup as I set up two SIP trunks, callcentric and simionics, and netiher one work for me.

Where do I configure NAT? When I define my trunks in SIP Protocol the General tab has a NAT field and I use NAT: yes (force rport + comedia) and then in Signaling I put Yes in Monitoring. Is there more to do beyond those two settings?

I am currently forwarding ports 5060-5069 and 5222 (I read google voice uses that port) and 100000-50000.

 

[billsimon]

Where do I configure NAT? When I define my trunks in SIP Protocol the General tab has a NAT field and I use NAT: yes (force rport + comedia) and then in Signaling I put Yes in Monitoring. Is there more to do beyond those two settings?

You need to define your local network(s) and public IP. I do not know where this is done in Wazo.

 

[Sylvain Boily]

In PBX->general config->SIP->network.

 

[rogerjbos]

I finally got a callcentric DID to work with good audio, but I had to put in my External IP address: in PBX->general config->SIP->network as mentioned above. I don't like this, because doesn't this mean whenever the dynamic IP address my ISP gives me changes my phones will stop working until I go into Wazo and update this setting? If so, that doesn't sound like a very reliable setup. Am I better off using a cloud environment?