ALERT GVSIP Is Dead, was: GVSIP Registration Issues. Asterisk 13.23.0 Seems To Fix The Problem.

[Marcus Watson]

Ok, I was able to resolve it by upgrading Asterisk using the instructions in post 13. Thanks!

 

[Vladimir Mikhelson]

Ok, I was able to resolve it by upgrading Asterisk using the instructions in post 13. Thanks!

Marcus,

Good move.

-Vladimir

 

[Vladimir Mikhelson]

Seems to be the common thread with this problem.

I don't want to hijack the thread, but why use those things? Processing power is limited, data storage is slow, and there's not real advantage to miniaturization or low power consumption for a pbx. You could beat the cost and outperform a Raspi by using an old Core2Duo box you could pick up for $50. What's the purpose in using one of those things for a pbx?

Ajonate,

Just my reasoning, also hoping not to derail the thread I opened to track another problem.

RPi is low power consumption, low heat emissions, can be hung on a wall, very stable, no moving parts to break. There are certain limitations of course.

-Vladimir

 

[ariban]

Ariban,

Is it still working for you? It looks like your problem where "amportal restart" was the only remedy to recover registrations started affecting me as of 3am Tuesday morning, 10/16/2018.

Still cannot find a solution.

-Vladimir

EDIT

After re-reading your messages more carefully I concluded your problem was slightly different. Your registrations showed as registered but did not actually work. In my case they become "rejected" after an hour or so after amportal restart.

-Vladimir

sorry i was traveling, no its not working for me.
well i saved a copy of my virtual file before making upgrade and changes. somehow my old file (i am using a very old gvsip folder and it works better than the newer one, so i went back to that one. at least incoming mostly works, but outgoing does not work until i amportal restart!

 

[Vladimir Mikhelson]

sorry i was traveling, no its not working for me.
well i saved a copy of my virtual file before making upgrade and changes. somehow my old file (i am using a very old gvsip folder and it works better than the newer one, so i went back to that one. at least incoming mostly works, but outgoing does not work until i amportal restart!

Ariban,

Thank you for the update. It soungs like your issue is very specific to your invirnment and needs more analysis on your part.

FYI, my latesst issue was a pretty wierd DNS issue. After it was fixed everything is back to normal on the latest Asterisk version.

-Vladimir

 

[BostonDan]

Ok, so I ran the script to set the new GV-SIP outbound proxy. I have no issues with my asterisk for connections internal to my network, but now my external softphone is not receiving any sip voice packets, whether the call originates from the softphone or the call originates from a standard cell phone.

I am using TLSV1 and the encryption is enabled.
I have checked my port forwarding and have my RTP ports properly forwarded from external to my network to my internal network. The call connects properly and will sit for minutes but no sip voice packets are being received or sent.
I have tried different codecs (g729, uLaw, aLaw, etc.) but same result.

I have downloaded and compiled asterisk 13.23.1 and all seems good, but this issue occurs with just external devices (I use Users and Devices on my setup to permit my home phone and cell phone's sip softphone to ring simultaneously).

Anybody have any ideas?

Cheers,
B.D.

 

[Vladimir Mikhelson]

Ok, so I ran the script to set the new GV-SIP outbound proxy. I have no issues with my asterisk for connections internal to my network, but now my external softphone is not receiving any sip voice packets, whether the call originates from the softphone or the call originates from a standard cell phone.

I am using TLSV1 and the encryption is enabled.
I have checked my port forwarding and have my RTP ports properly forwarded from external to my network to my internal network. The call connects properly and will sit for minutes but no sip voice packets are being received or sent.
I have tried different codecs (g729, uLaw, aLaw, etc.) but same result.

I have downloaded and compiled asterisk 13.23.1 and all seems good, but this issue occurs with just external devices (I use Users and Devices on my setup to permit my home phone and cell phone's sip softphone to ring simultaneously).

Anybody have any ideas?

Cheers,
B.D.

BostonDan,

Can you please confirm the following:
1. Is soft phone using SIP, PJSIP, or IAX?
2. How is cell phone talking to Asterisk?
3. Does GVSIP trunk show "Registered"?

-Vladimir

 

[BostonDan]

BostonDan,

Can you please confirm the following:
1. Is soft phone using SIP, PJSIP, or IAX?
2. How is cell phone talking to Asterisk?
3. Does GVSIP trunk show "Registered"?

-Vladimir

Vladimir,

I use SIP on my cell phone. The Cell phone has Zoiper pro on it, and it was working perfectly before the update to the GV-SIP outbound proxy change.
The GVSIP trunk (all 4, I have 2 home lines + 1 fax line for my home, plus a fourth line only on my cell phone that I register to my cell phone) shows registered.

Thank you for any insight.

Cheers,
B.D.

 

[Vladimir Mikhelson]

Vladimir,

I use SIP on my cell phone. The Cell phone has Zoiper pro on it, and it was working perfectly before the update to the GV-SIP outbound proxy change.
The GVSIP trunk (all 4, I have 2 home lines + 1 fax line for my home, plus a fourth line only on my cell phone that I register to my cell phone) shows registered.

Thank you for any insight.

Cheers,
B.D.

BostonDan,

Sorry, I am still not sure I understand your layout / problem completely. Please correct me anywhere I am wrong.
1. Zoiper on your cell is registered as an extension to your Asterisk via SIP
2. Zoiper used to be able to place and receive calls, but after you have installed GVSIP on your Asterisk it fails in both directions
3. Four (4) GVSIP trunks are registered and work as expected on your Asterisk, except no connectivity with your Zoiper

If all of the above correct I would look into Zoiper's STUN setting and try to play with these. GVSIP actively uses STUN, and I can see a potential conflict.

-Vladimir

EDIT.

I just looked into the iPhone Zoiper settings, and discovered they removed majority of the settings which used to be available in the desktop version I had had experience with including STUN settings.

Since Zoiper does not offer advanced settings any more I should say your best bet is to capture the traffic by tcpdump on your Asterisk server and / or on your Internet router / gateway and check where the Zoiper packets are being sent, specifically, to a private or to a public IP. Assuming your Zoiper is on the provider's network, not on the WiFi, the address should be public. In case it is on a LAN and traffic is still being sent to a public IP then your router / gateway should be set to provide a loopback or Asterisk needs to be adjusted in the sip.conf as far as local networks go.

Something along these lines.

-Vladimir

 

[BostonDan]

BostonDan,

Sorry, I am still not sure I understand your layout / problem completely. Please correct me anywhere I am wrong.
1. Zoiper on your cell is registered as an extension to your Asterisk via SIP
2. Zoiper used to be able to place and receive calls, but after you have installed GVSIP on your Asterisk it fails in both directions
3. Four (4) GVSIP trunks are registered and work as expected on your Asterisk, except no connectivity with your Zoiper

If all of the above correct I would look into Zoiper's STUN setting and try to play with these. GVSIP actively uses STUN, and I can see a potential conflict.

-Vladimir

Vladimir,

Vladimir,

You are close. Step 1 is correct. Step 2, I can make and receive calls, but there is no sound on either end (originator or destination).
Step 3 is incorrect as well, as the GVSIP trunks are registered and work both internally (CISCO SPA112 Analog Telephone Adapter). and externally. Calls are connected, but externally there is no sound on either end, yet internally the calls have 2-way audio and work as expected for both originating and receiving calls.

I will try STUN settings. I had left these off previously for the external Zoiper softphone and it worked with two-way audio under the previous outbound proxy for GVSIP (obihai.telephony.goog:5061).

Cheers,
Dan

 

[Vladimir Mikhelson]

Vladimir,

Vladimir,

You are close. Step 1 is correct. Step 2, I can make and receive calls, but there is no sound on either end (originator or destination).
Step 3 is incorrect as well, as the GVSIP trunks are registered and work both internally (CISCO SPA112 Analog Telephone Adapter). and externally. Calls are connected, but externally there is no sound on either end, yet internally the calls have 2-way audio and work as expected for both originating and receiving calls.

I will try STUN settings. I had left these off previously for the external Zoiper softphone and it worked with two-way audio under the previous outbound proxy for GVSIP (obihai.telephony.goog:5061).

Cheers,
Dan

Dan,

One-way audio or no audio with SIP is either NAT or firewall issue or both. I would capture the traffic, specifically RTP, verify the destination IP. Debug on Asterisk can show you the details of RTP negotiation, although you mentioned uLaw so unsuccessful negotiation should not be a problem.

One more little "red flag" is TLS1.1. There was a speculation on DSLReports that the switch to voice.telephony.goog required TLS1.2 support. But on the other hand, should that be the case then your GVSIP trunks would have not registered. And also it should have nothing to do with the SIP audio issue.

-Vladimir

 

[mcfuzz]

Hi all,

Recently I've started getting registration rejected messages on my iPBX 13-13.7 running on Scientific Linux 6.9 (i.e. I did the ISO deployment on a VM server).

<Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 gvsip1/sip:obihai.sip.google.com                        gvsip1            Rejected

Objects found: 1

Refreshed my refresh token - still does not help. Ensured my pjsip_custom.conf is updated with the proper addresses:

outbound_auth=gvsip1
server_uri=sip:eek:obihai.sip.google.com
outbound_proxy=sip:voice.telephony.goog:5061\;transport=tls\;lr\;hide

Not sure where to proceed from here

Interestingly enough - when I first log in and run the update utility - it says there are updates available as of today, but it seems like everything is installed...?

Retrieving... lastupdate7
Updates available as of Sat Nov 10 12:41:12 PST 2018: 30
Checking for update71. INSTALLED: Firewall startup reconfigured
Checking for update72. INSTALLED: status update for SL7
Checking for update73. INSTALLED: IPtables boot sequence patch
Checking for update74. INSTALLED: BASH Vulnerability patch
Checking for update75. INSTALLED: BASH #2 Vulnerability patch
Checking for update76. INSTALLED: FreePBX ARI Vulnerability patch
Checking for update77. INSTALLED: FreePBX ARI #2 Vulnerability patch
Checking for update78. INSTALLED: Incredible PBX LAN Vulnerability Patches
Checking for update79. INSTALLED: FreePBX Web Access Patch
Checking for update710. INSTALLED: rc.local executable patch
Checking for update711. INSTALLED: Fixed Weaather by ZIP Code TTS script
Checking for update712. INSTALLED: Fixed Weaather by ZIP Code TTS script
Checking for update713. INSTALLED: IPtables security patch applied
Checking for update714. INSTALLED: IPtables security patch applied
Checking for update715. INSTALLED: ConfigEdit patch applied
Checking for update716. INSTALLED: Fail2Ban patch applied
Checking for update717. INSTALLED: Asterisk logroate patch applied
Checking for update718. INSTALLED:
Checking for update719. INSTALLED: Outbound calling security patch
Checking for update720. INSTALLED: Outbound calling security patch #2
Checking for update721. INSTALLED: Outbound calling security patch #3
Checking for update722. INSTALLED: rc.local startup file enabled.
Checking for update723. INSTALLED: CentOS/SL 7 iptables-restart fix
Checking for update724. INSTALLED: PBX status fix for Public IP address
Checking for update725. INSTALLED: Asterisk log rotate patch applied
Checking for update726. INSTALLED: ipchecker patch for TM3 applied
Checking for update727. INSTALLED: Asterisk running as root user patch
Checking for update728. INSTALLED: nv-weather-zip updated for NWS access
Checking for update729. INSTALLED: hostname updated for SendMail on HiFormance
Checking for update730. INSTALLED: GVSIP updated for Google *improvement*
Updates and notifications completed.

One thing to note - when I do fwconsole/amportal restart, it complains about not having a DAHDI module - presumably that is not needed for gvsip, correct?

Any help is appreciated

 

[Dobs14]

Hi all,

Recently I've started getting registration rejected messages on my iPBX 13-13.7 running on Scientific Linux 6.9 (i.e. I did the ISO deployment on a VM server).

<Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 gvsip1/sip:obihai.sip.google.com                        gvsip1            Rejected

Objects found: 1

Refreshed my refresh token - still does not help. Ensured my pjsip_custom.conf is updated with the proper addresses:

outbound_auth=gvsip1
server_uri=sip:eek:obihai.sip.google.com
outbound_proxy=sip:voice.telephony.goog:5061\;transport=tls\;lr\;hide

Not sure where to proceed from here

Interestingly enough - when I first log in and run the update utility - it says there are updates available as of today, but it seems like everything is installed...?

Retrieving... lastupdate7
Updates available as of Sat Nov 10 12:41:12 PST 2018: 30
Checking for update71. INSTALLED: Firewall startup reconfigured
Checking for update72. INSTALLED: status update for SL7
Checking for update73. INSTALLED: IPtables boot sequence patch
Checking for update74. INSTALLED: BASH Vulnerability patch
Checking for update75. INSTALLED: BASH #2 Vulnerability patch
Checking for update76. INSTALLED: FreePBX ARI Vulnerability patch
Checking for update77. INSTALLED: FreePBX ARI #2 Vulnerability patch
Checking for update78. INSTALLED: Incredible PBX LAN Vulnerability Patches
Checking for update79. INSTALLED: FreePBX Web Access Patch
Checking for update710. INSTALLED: rc.local executable patch
Checking for update711. INSTALLED: Fixed Weaather by ZIP Code TTS script
Checking for update712. INSTALLED: Fixed Weaather by ZIP Code TTS script
Checking for update713. INSTALLED: IPtables security patch applied
Checking for update714. INSTALLED: IPtables security patch applied
Checking for update715. INSTALLED: ConfigEdit patch applied
Checking for update716. INSTALLED: Fail2Ban patch applied
Checking for update717. INSTALLED: Asterisk logroate patch applied
Checking for update718. INSTALLED:
Checking for update719. INSTALLED: Outbound calling security patch
Checking for update720. INSTALLED: Outbound calling security patch #2
Checking for update721. INSTALLED: Outbound calling security patch #3
Checking for update722. INSTALLED: rc.local startup file enabled.
Checking for update723. INSTALLED: CentOS/SL 7 iptables-restart fix
Checking for update724. INSTALLED: PBX status fix for Public IP address
Checking for update725. INSTALLED: Asterisk log rotate patch applied
Checking for update726. INSTALLED: ipchecker patch for TM3 applied
Checking for update727. INSTALLED: Asterisk running as root user patch
Checking for update728. INSTALLED: nv-weather-zip updated for NWS access
Checking for update729. INSTALLED: hostname updated for SendMail on HiFormance
Checking for update730. INSTALLED: GVSIP updated for Google *improvement*
Updates and notifications completed.

One thing to note - when I do fwconsole/amportal restart, it complains about not having a DAHDI module - presumably that is not needed for gvsip, correct?

Any help is appreciated

Please see https://pbxinaflash.com/community/threads/problems-with-gv-again.23283/

To sum up, run:
sed -i 's|voice.telephony.goog|64.9.242.172|g' /etc/asterisk/pjsip_custom.conf
sed -i 's|voice.telephony.goog|64.9.242.172|g' /root/gvsip-naf/install-gvsip
amportal restart

 

[mcfuzz]

Please see https://pbxinaflash.com/community/threads/problems-with-gv-again.23283/

To sum up, run:
sed -i 's|voice.telephony.goog|64.9.242.172|g' /etc/asterisk/pjsip_custom.conf
sed -i 's|voice.telephony.goog|64.9.242.172|g' /root/gvsip-naf/install-gvsip
amportal restart

Thanks!! That did the trick

 

[shetu]

Please see https://pbxinaflash.com/community/threads/problems-with-gv-again.23283/

To sum up, run:
sed -i 's|voice.telephony.goog|64.9.242.172|g' /etc/asterisk/pjsip_custom.conf
sed -i 's|voice.telephony.goog|64.9.242.172|g' /root/gvsip-naf/install-gvsip
amportal restart

Thanks. It works for me too.

 

[Marcus Watson]

I had done this previously, but after changing to wifi on my Pi device, I had to do it again. Do I have to run these commands any time I restart the server?

 

[Vladimir Mikhelson]

I had done this previously, but after changing to wifi on my Pi device, I had to do it again. Do I have to run these commands any time I restart the server?

Marcus,

The answer is, "No." You will only need to change the settings if Google changes something on their end again.

-Vladimir

 

[AgentJeffy]

Here's the fix:

sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /etc/asterisk/pjsip_custom.conf
amportal restart
sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /root/gvsip-naf/install-gvsip

How long should the first command take to execute? I type it into my console and it just brings up a new line with no CD shown and seems to hang there. I'm using IncrediblePBX 13-13 but have upgraded Asterisk to 13.23.1 but I'm sure it was hanging on the older version of Asterisk as well.

Edit 1:
Okay let me correct that statement. I was using ` instead of ' at the beginning of 's|obihai.telephony... once corrected the command seems to execute very quickly but I'm still showing unregistered under my Chan_PJSip Registrations. Do I need to readd the gvsip after doing this fix?

Edit 2:
I'm going to reinstall IncrediblePBX 13-13 from scratch with no addons this time and see if I can get it working out of the gate, going to follow the instructions from Nerdvittles again and hope for the best...

 

[AgentJeffy]

Okay so I'm at a loss here. I did a clean install of IncrediblePBX 13-13 the full enchilada (it says the version number is 13-13.7) and then I applied the fix from #64 and then added the GVSIP via the console and nothing. Then I tried applying the fix again AFTER I had added the GVSIP and it's still telling me that I'm not registered so I have no idea what I'm doing wrong...

Edit 1:
Okay, so I don't know what I did differently this time but I finally got it registered. I'm at a loss here but I sure hope they don't change the address again. I think if you're on a FRESH install you have to follow the instructions here: http://nerdvittles.com/?p=26315

Just make sure you do the OCTOBER update first and then the November. Seemed to work for me. So I don't know. Anyways, thanks for the discussions here folks.

 

[Dobs14]

Okay so I'm at a loss here. I did a clean install of IncrediblePBX 13-13 the full enchilada (it says the version number is 13-13.7) and then I applied the fix from #64 and then added the GVSIP via the console and nothing. Then I tried applying the fix again AFTER I had added the GVSIP and it's still telling me that I'm not registered so I have no idea what I'm doing wrong...

What are both occurrences of "outbound_proxy=" set to in /etc/asterisk/pjsip_custom.conf?

You need to make sure the installer is changed too:

sed -i 's|voice.telephony.goog|64.9.242.172|g' /etc/asterisk/pjsip_custom.conf
sed -i 's|voice.telephony.goog|64.9.242.172|g' /root/gvsip-naf/install-gvsip
amportal restart