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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04

asterisk
pjsip
freepbx
Tags: #<Tag:0x00007fdfa08c3340> #<Tag:0x00007fdfa08c31d8> #<Tag:0x00007fdfa08c3070>

(Itzik) #316

I don’t think this issue is related to this particular setup. Seems more like a general network/registration issue.

Lot’s of people here are complaining on Cisco phones with FreePBX. For the sake of testing, you can try registering a (free) softphone like Zoiper or X-Lite and see if you are still having registration issues.

If you do, I think you should create a new topic. :slight_smile:

Much luck.


(GPz1100) #317

@xekon

Do you maintain a change log of how this guide has changed evolved over time? I visited this thread about 4 weeks ago, then got side tracked with other stuff. Back now. :slight_smile:


(Itzik) #318

If you click on the little pencil on the top right you can see each version of his 156 post edits. However, it does not seem like he added a brief description what was changed.


#319

Originally I did not have pictures for the GUI setup, also I did not have the companion thread for setting up Oauth credentials initially.

Not a whole lot changed, mostly just formatting changes to make the guide easier to follow.

OH! and this did not tightly integrate with FreePBX originally. more of it was in the _custom file, Thanks to Billsimon and his recommendations, some of what was in the _custom file such as the dial plan and routes, are now able to be done in the GUI.


#320

It does sounds like a networking issue… Is the server your using a computer on your local network (LAN) or is it a virtual machine or a server hosted elsewhere.

I have also tested Csipsimple softphone, and that works for me as well, as a softphone to test with.

I am not sure what your experience level is with linux/computers, etc. If you think you might have overlooked something in the logs I would be happy to take a look. Issue a reboot, then attempt a few calls from both 701 and 702 (I know they wont work, but it should show the activity, of what they are trying to do, hopefully)


(Itzik) #322

Default (pjsip) port is 5060 unless you set it to 6050


#323

@ohaydel I sent you a PM, I do not even see the registration attempt from 701/702 in that log.

I will need you to either email me the full log (following a reboot)

or

you can create a pastebin post of the full log, set it to unlisted then post the link here or PM me the link.

If you post the link here then more eyes can look at it to try and help, but if you are not sanitizing any of the fields then you may prefer to PM it instead.


(Ohaydel) #324

Was a typo on my part. Same results with 5060.


#325

To look at the section of the log that you did post, it looks to me like the google voice trunk is registering just fine.

What I do not see is any registration attempts from your ATA devices / softphones, as if they are not configured to point to your asterisk machine (192.168.1.18) or something… or you just didnt post that part of the log?


#326

That is interesting, I definitely do NOT get that error, filtering my log for AMI returns only this:

[2018-07-29 13:29:13] VERBOSE[3854] loader.c: Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support)

#327

I thought of something here @ohaydel so basically when you copy and paste from windows on putty to linux, like when you use nano to edit the file.

windows uses different type of carriage return CRLF, linux only uses LF, this can cause issues sometimes.

what you can do is use a program called dos2unix to correct the files:

sudo apt-get install dos2unix

sudo dos2unix /etc/asterisk/pjsip_custom_post.conf

This might solve your problem, or not, with the information that I have so far it is hard to say exactly what the issue is.


(Ohaydel) #328

I was pulling the log from the WebUI into the FreeBbx server. Copy and paste. That was everything that appeared after a reboot, and after I restarted the phone.


(Ohaydel) #329

Yea, they give no indication that they are connecting to the server. Let me see if I can paste a photo of my config. I have the inbound and outbound proxies set to 192.168.1.18:5060. All I have filled out is the UseriD: 701 and the password: (The secret)

!


(Ohaydel) #330


#331

Try 5160


#332

on Putty do this:

sudo asterisk -rvvv

then pickup your phones and try placing a call, this should force them to interact with the server, such as:

<--- Transmitting SIP request (423 bytes) to UDP:192.168.5.156:5160 --->
OPTIONS sip:5003@192.168.5.156:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.240:5060;rport;branch=z9hG4bKPj0804f59f-fd8f-4535-a73b-b8b74ba6023a
From: <sip:5003@192.168.5.240>;tag=806c24b7-f7c4-41a9-897a-3b907567db16
To: <sip:5003@192.168.5.156>
Contact: <sip:5003@192.168.5.240:5060>
Call-ID: 006f733f-fa6c-4082-9af6-c2c9dd79d0f9
CSeq: 62709 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.3.6(15)
Content-Length:  0


<--- Received SIP response (448 bytes) from UDP:192.168.5.156:5160 --->
SIP/2.0 200 OK
To: <sip:5003@192.168.5.156>;tag=791bc7843cbe71b5i0
From: <sip:5003@192.168.5.240>;tag=806c24b7-f7c4-41a9-897a-3b907567db16
Call-ID: 006f733f-fa6c-4082-9af6-c2c9dd79d0f9
CSeq: 62709 OPTIONS
Via: SIP/2.0/UDP 192.168.5.240:5060;branch=z9hG4bKPj0804f59f-fd8f-4535-a73b-b8b74ba6023a
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

share the output with us, maybe it will give clues.


#333

Set BOTH the userID and AuthID to 701, atleast that is how I have my extensions configured, and set use AuthID to yes.

Also I can see “use auth invite: no” and Anonymouse “yes”, I cannot say for sure, but I do not think that will work like that, you likely need the auth invite.


#334

definitely do more testing with CSipSimple, worry about your other phone later.

In CSipSimple, its very SIMPLE!

You create a “Basic” type account, toward the bottom, lt will prompt for four fields

Account name: 701
User: 701
Server: 192.168.1.18:5160
Password: passwordhere

The reason we wanna try CSipSimple first, is that its simple, and so we can rule out the phone configuration as being your primary issue.

I would run:

sudo asterisk -rvvv

then use CSipSimple to try placing a few calls, if your not seeing any output in the asterisk CLI (sudo asterisk -rvvv) then something is VERY wrong with your install, and without being there to look things over I cannot say for sure what the problem might be, you may have to try installing from scratch again if your seeing no related output in the Asterisk CLI while trying to place calls.

your CRITICAL AMI error for instance, is different from any of the install I have done, I have never seen that error.

Once you have CSipSimple working, then we can move on to trying to get your other phone working.


(User permanently banned) #335

Recently I had an issue settying up some cisco phones on Asterisk 13

Asterisk refused to see anything sent by these phones till the NAT mode was changed. Try changing NAT mode in extension set up


(Ohaydel) #336

I have it Yes and Yes right now on the Cisco, is yours set to No?