ALERT GV: The Sky Has Fallen... Really

wardmundy

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I guess my thinking is, why be a luddite and make users go thru that unnecessary interruption to their business conversation and thought flow, due to an unnecessary technical difficulties interruption. This PBX system is equipped with an advanced software stack specially designed to perform fast relatively seamless handover to a new IP address, in about a second, even when the user's endpoint suddenly auto logs on to a WiFi access point behind a NAT such as you find at all coffee shops retail stores public places of interest etc.

I was just kidding although your luddite assessment is fairly accurate. :red indian:
 

chris_c_

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Ring groups not working, blfs not working, general weird behavior - call parking/pickup issues if I recall, etc. Some things like the ring group issues definitely got better in FPBX13, we at least still had some weird blf issues IIRC. Had two smaller locations we tried starting out as pure PJSIP(it's the future after all) - that didn't last long. Not all of this (probably most of it) were technically PJSIP issues, more likely FPBX, (and maybe some of our own config), but that doesn't matter, there's no need to waste hours debugging when chan_sip just works. I'm perfectly willing to wait and let others get it worked out.
It's best to get your issues looked into and assessed, and if truly bug(s), fixed.
If any of your reports are truly bugs in PJSIP, then you'll help not only yourself but also the whole community of PJSIP and PBX users by doing your part to help get them fixed sooner rather than later.
So you should report these issues to the PJSIP developers on the PJSIP mailing list here:
https://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
 
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billsimon

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For instance, you're on call, on your VOIP smartphone app, on office WiFii data. You walk outside, stil talking on call, on 4G LTE. You keep talking and walking, and arrive at the local starbucks or burger pub for lunch, your smart phone auto connects and logs into the free WiFi there, which you like because it's a rock solid 50 megabit connection. AND it's got a typical WiFI NAT gateway router. With a commonly available smart phone, VOIP app, it's easy to demonstrate the call drops when there's no ICE STUN TURN and IP6. These newer technologies are needed to get RTP UDP audio packets and SIP signalling thru many NAT gateways, especially when you went from not having a NAT gateway, then you walk to a place that has a NAT gateway, and the call is still ongoing. Why take the chance of letting those walking smart phone VOIP app users' calls drop, interrupt business, and lose money and time? Letting their calls continue by enabling ICE STUN TURN and IP6 is the cleanest safest cheapest and smartest way to deal with this walking around and getting a new IP on a WiFi behind a NAT gateway which you don't own or control yet still want to continue your calls and win business in all scenarios IMHO.

Would you mind writing a post (in a separate thread, as the topic is unrelated to this thread's topic) on how exactly to set up this smooth hand-off behavior? I can't tell whether you're describing an actual configuration you have done or a hypothetical, but I haven't seen any documentation on how to set up this kind of hand-off with Asterisk/FreePBX.
 

chris_c_

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Would you mind writing a post (in a separate thread, as the topic is unrelated to this thread's topic) on how exactly to set up this smooth hand-off behavior? I can't tell whether you're describing an actual configuration you have done or a hypothetical, but I haven't seen any documentation on how to set up this kind of hand-off with Asterisk/FreePBX.

MID-CALL MOBILITY aka HANDOVER, HANDOFF
https://pbxinaflash.com/community/threads/mid-call-mobility-aka-handover-handoff.23053/
 

troysmoke

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Good Afternoon,

I ran the NAF update again last night. I've been running into an issue where the GV trunk will be up initially (I can call out on a softphone), but then it goes down and looks like it never comes back up.

Any thoughts? Do I need to rebuild again? I'm running a Raspberry Pi with the image from a week ago or so.

Thanks, troy
 

wardmundy

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This only affects those that wish to use PJSIP and TLS with extensions on your PBX. If so, you should apply the following patch which has been integrated into fresh installs of GVSIP moving forward. Rebuilding GVSIP on your PBX using the update procedure now includes this patch as well.

Code:
cd /etc/asterisk
sed -i 's|bind=0.0.0.0:5061|bind=0.0.0.0:5062|' pjsip_custom.conf
amportal restart
 
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windpoint

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I deployed the incrediblePBX in Hiformance. It has been running perfect. I switched the ISP yesterday and lost connection because the IP changed. I try to login via ssh with no luck.
Is there any way I can log into it if I dont have another whitelisted IP? Thank you.
 

shetu

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I deployed the incrediblePBX in Hiformance. It has been running perfect. I switched the ISP yesterday and lost connection because the IP changed. I try to login via ssh with no luck.
Is there any way I can log into it if I dont have another whitelisted IP? Thank you.
Find serial console In your vps control panel and log in vps via puty or java shell?
vps%20login.jpg
 

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tycho

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I deployed the incrediblePBX in Hiformance. It has been running perfect. I switched the ISP yesterday and lost connection because the IP changed. I try to login via ssh with no luck.
Is there any way I can log into it if I dont have another whitelisted IP? Thank you.

I would guess that the "port knock" program was installed (part of "travelin' man?") when you deployed IPBX. When first launched, your screens told you the "knock ports" set for you, and how to get in.

Get yourself a "port knocking" program, do the "knocking," and get in.

See here (http://nerdvittles.com/?p=9871) and here (http://nerdvittles.com/?p=24058#app8) for background
 

chris_c_

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As @shetu and @tycho say above:

1. Did you make note of the 3 randomly selected port numbers, for you to use in a port knocker smart phone app or desktop app, which the installer displayed to you at the end of the installation script ?

2. If you didn't make note of your unique 3 port knocker ports, then your only way to log in is thru your VPS provider's VPS control panel (virtualizor or solusvm or other) click on "Serial Console", then either use the temporary credentials it displays for login with your PuTTY ssh app, or click on "HTML5 console" to access the serial console terminal in your web browser.
 

windpoint

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Find serial console In your vps control panel and log in vps via puty or java shell?
vps%20login.jpg


I was able to get the new IP whitelisted. Thank you.

from your screenshot, there is a "reinstall" button. How did you get this option out? I dont see that in my Hiformance portal.

@tycho, good information. will look into it. Thank you.
 

shetu

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I was able to get the new IP whitelisted. Thank you.

from your screenshot, there is a "reinstall" button. How did you get this option out? I dont see that in my Hiformance portal.

@tycho, good information. will look into it. Thank you.
I am using woothosting.com vps. It has separate control panel. May be your vps website has another design.
 
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troysmoke

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Looks like this thread has quieted down. Must mean things are working well.

I haven't been able to successfully make calls for a while now. I just rebuilt today using the latest image, got new refresh tokens, etc.

The trunks connect to Google, but every time I attempt to make a call, I get a:
[2018-09-23 17:03:07] DEBUG[15348] res_pjsip_outbound_registration.c: Setting transport to 0x75d615d4
[2018-09-23 17:03:07] DEBUG[15348] res_pjsip.c: Overriding endpoint transport to use 0x75d615d4
[2018-09-23 17:03:07] VERBOSE[15395][C-0000000f] app_dial.c: Called PJSIP/1NXXNXXXXXX@gvsip1
[2018-09-23 17:03:07] DEBUG[15348] res_pjsip_outbound_registration.c: Found matching outbound registration state
[2018-09-23 17:03:07] DEBUG[15348] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:[removed]>
[2018-09-23 17:03:07] DEBUG[15348] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:[removed[>
[2018-09-23 17:03:07] VERBOSE[15395][C-0000000f] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)
[2018-09-23 17:03:07] VERBOSE[15395][C-0000000f] pbx.c: Executing [s@macro-dialout-trunk:32] NoOp("SIP/701-0000000c", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack

Update:
If I reboot the Pi, I can get a call out. But, it soon fails with the following error in the logs:
Error sending STUN request: Network is unreachable

Update 2:
After a bit of time, these error messages show up as well:
[2018-09-23 17:30:46] WARNING[1901] pjproject: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2018-09-23 17:30:54] DEBUG[1901] res_pjsip_outbound_registration.c: Registration using newly created transport 0x75d7b344
[2018-09-23 17:31:22] WARNING[1901] pjproject: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2018-09-23 17:31:22] DEBUG[3700] res_pjsip_outbound_registration.c: Registration using newly created transport 0x75926c64
[2018-09-23 17:40:55] WARNING[1901] pjproject: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2018-09-23 17:41:02] DEBUG[1901] res_pjsip_outbound_registration.c: Registration using newly created transport 0x75d9465c
[2018-09-23 17:41:24] WARNING[1901] pjproject: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2018-09-23 17:41:24] DEBUG[4568] res_pjsip_outbound_registration.c: Registration using newly created transport 0x75d9e20c
[2018-09-23 17:51:03] WARNING[1901] pjproject: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2018-09-23 17:51:14] ERROR[1901] res_pjsip_outbound_registration.c: Resolver failed. Cannot send message
[2018-09-23 17:51:25] WARNING[1901] pjproject: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2018-09-23 17:51:25] ERROR[5364] res_pjsip_outbound_registration.c: Resolver failed. Cannot send message

Thoughts? As far as I can tell, the trunks are connected to Google, but no calls are getting through.

Any help would be greatly appreciated. zoddoz
 
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chris_c_

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Code:
[2018-09-23 17:51:25] ERROR[5364] res_pjsip_outbound_registration.c: Resolver failed. Cannot send message

Thoughts? As far as I can tell, the trunks are connected to Google, but no calls are getting through.

Any help would be greatly appreciated.

1. What makes you say, “As far as I can tell, the trunks are connected to google.”?? This is not a rhetorical question. What proof do you have the trunks are connected to google. Apologies if this sounds off, it’s not intended to! Just asking how you know since you didn’t explain how you knew in your question.

2. What’s the output of these commands:
Code:
ping yahoo.com
ping stun.counterpath.net

EDIT:
3. Have you tried upgrading to Asterisk 13.23 ??
See here for how:
https://pbxinaflash.com/community/threads/gvsip-registration-issues-asterisk-13-23-0-seems-to-fix-the-problem.23083/
 
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troysmoke

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1. What makes you say, “As far as I can tell, the trunks are connected to google.”?? This is not a rhetorical question. What proof do you have the trunks are connected to google. Apologies if this sounds off, it’s not intended to! Just asking how you know since you didn’t explain how you knew in your question.

2. What’s the output of these commands:
Code:
ping yahoo.com
ping stun.counterpath.net

EDIT:
3. Have you tried upgrading to Asterisk 13.23 ??
See here for how:
https://pbxinaflash.com/community/threads/gvsip-registration-issues-asterisk-13-23-0-seems-to-fix-the-problem.23083/
"As far as I can tell", (cause I'm an Asterisk newb) the logs appear to show a successful connection, and the OBiTALK devices show up in Google, so the trunk registration works.

Pings succeed to both locations.

Thanks, I will give the upgrade a try.

zoddoz
 

wardmundy

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Looks like Google Voice GVSIP is dead again this morning.Trunks register but all calls in and out fail. Probably time to find a reputable provider.
 

mbellot

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Looks like Google Voice GVSIP is dead again this morning.Trunks register but all calls in and out fail. Probably time to find a reputable provider.

That time was several years ago... I never saw the value, too much time and tinkering for something that would stop working.

Voip.ms has been rock solid for so many years I've lost count, the $7 (approximate) per month I spent on service is worth the peace of mind.

Besides, there was always something that didn't sit right with me having Google in the middle of all my phone conversations.
 

HermanMiller

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Looks like Google Voice GVSIP is dead again this morning.Trunks register but all calls in and out fail. Probably time to find a reputable provider.
No problems here.

I've had this seem to happen twice over the last month/month and a half - asterisk failing to complete calls, in or out, and asterisk -rvvvv showing 'congestion' but a simple server reboot fixes it (probably restarting Asterisk would be fine but I just reboot the whole box).
 

kdthomas

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No issues with my Google Voice trunks (3) today or for the past week. When they do go out, it's usually in the early AM for about 30 minutes or so. Maybe this is when Google is doing maintenance? I knew all along that anything you get for free comes with the unknown and I'm fine with that. I don't use this system for anything mission critical. It's just nice having a home phone that also rings my cell phone and vice versa so I can put up with an outage here or there.
 

kenn10

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No problems with my two GV lines. I use them for faxes and they're doing OK. I'm sure they'll die before long.
 

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