IncrediblePBX and GVSIP registration issues

Mike De

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Hi, this started a week or so ago and messed with it a little but haven't gotten anywhere. I had 3 GVSIP trunks working fine on 2 different IncrediblePBX VM images for testing purposes working fine for at least a month. Now they don't stay Registered anymore, I don't have an exact time frame, maybe an hour?

pjsip show registrations

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

gvsip1/sip:eek:bihai.sip.google.com gvsip1 Rejected
gvsip2/sip:eek:bihai.sip.google.com gvsip2 Rejected

If i run pjsip send register gvsip1 it'll sometimes get them to come back as Registered. or sometimes i get the following:
[2018-09-30 17:31:56] ERROR[11192]: res_pjsip_outbound_registration.c:598 stateless_send_resolver_callback: Resolver failed. Cannot send message
[2018-09-30 17:31:56] ERROR[11192]: res_pjsip_outbound_registration.c:598 stateless_send_resolver_callback: Resolver failed. Cannot send message

DNS works fine on them, and this is the same results on 2 totally different VMs.

I've had pjsip set logger on but doesn't tell me much other than sometimes seeing:

res_pjsip_outbound_registration.c:989 schedule_retry: No response received from 'sip:eek:bihai.sip.google.com' on registration attempt to 'sip:[email protected]', retrying in '60'

I can do a fwconsole restart and that'll bring them both back up to Registered, but they don't stay like that, they eventually always go back to Rejected.
 

Mike De

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I would delete the GVSIP trunks then add them back.

**Edit

As I look at mine in the Chan_PJSip Channel(s) section I see

No objects found.

Below that in section Chan_PJSip Registrations I see the registered trunks

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

 gvsip1/sip:obihai.sip.google.com                        gvsip1            Registered     
 gvsip2/sip:obihai.sip.google.com                        gvsip2            Registered     

Objects found: 2




 
Last edited:

Mike De

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Had a fresh install with the patch running fine for about a week. I woke up this morning and found both trunks are rejected. if i try resending the registration I get the following.


IncrediblePBX*CLI> pjsip send register gvsip2
[2018-10-12 09:03:12] DEBUG[54479]: res_pjsip_outbound_registration.c:625 stateless_send_resolver_callback: Registration using newly created transport 0x7f1c302e2658
[2018-10-12 09:03:12] WARNING[45873]: pjproject:0 <?>: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336151568> <SSL routines-ssl3_read_bytes-sslv3 alert handshake failure> len: 0
[2018-10-12 09:03:12] WARNING[8764]: res_pjsip_outbound_registration.c:987 schedule_retry: No response received from 'sip:eek:bihai.sip.google.com' on registration attempt to 'sip:[email protected]', retrying in '60'
 

ajonate

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Same here.

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

gvsip1/sip:eek:bihai.sip.google.com gvsip1 Rejected
gvsip2/sip:eek:bihai.sip.google.com gvsip2 Rejected
gvsip3/sip:eek:bihai.sip.google.com gvsip3 Rejected

Objects found: 3
 

ajonate

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OK, a solution has been posted and it's working for me. Open the /etc/asterisk/pjsip_custom.conf file. Change the outbound proxy address from obihai.telephony.goog to voice.telephony.goog. There are two such entries for each gvsip trunk.

Find this:

outbound_proxy=sip:eek:bihai.telephony.goog:5061\;transport=tls\;lr\;hide

Change to:

outbound_proxy=sip:voice.telephony.goog:5061\;transport=tls\;lr\;hide

Then restart asterisk:

amportal restart
 
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ajonate

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Last edited:

rmlworld

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Thanks for the fix. I made the change to the pjsip_custom.conf this morning and I am back in business.
 

ws2000

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Thank you for the fix. Worked as advertised. Saved me hours of waisted time.
 

Laurence

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Alas didn't work for me. When I looked at /etc/asterisk/pjsip_custom.conf the lines are already

outbound_proxy=sip:voice.telephony.goog:5061\;transport=tls\;lr

yet when I start Asterisk and try to make a call it still references

[2018-11-12 02:33:30] WARNING[2626]: res_pjsip_outbound_registration.c:866 schedule_retry: No response received from 'sip:eek:bihai.sip.google.com' on registration attempt to 'sip:[email protected]', retrying in '60'

I have rebooted the Pi and restarted asterisk a number of times
 

ajonate

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Alas didn't work for me. When I looked at /etc/asterisk/pjsip_custom.conf the lines are already

outbound_proxy=sip:voice.telephony.goog:5061\;transport=tls\;lr

yet when I start Asterisk and try to make a call it still references

[2018-11-12 02:33:30] WARNING[2626]: res_pjsip_outbound_registration.c:866 schedule_retry: No response received from 'sip:eek:bihai.sip.google.com' on registration attempt to 'sip:[email protected]', retrying in '60'

I have rebooted the Pi and restarted asterisk a number of times

Google changed things again since that thread. Here is the thread for the new fix,

https://pbxinaflash.com/community/threads/problems-with-gv-again.23283/page-2

We're using the IP address now. Basically, the new fix is to change voice.telephony.goog to 64.9.242.172.

Here are the commands for the new fix.

cp /etc/asterisk/pjsip_custom.conf /etc/asterisk/pjsip_custom.conf.old
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
 
Last edited:

Laurence

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Google changed things again since that thread. Here is the thread for the new fix,

https://pbxinaflash.com/community/threads/problems-with-gv-again.23283/page-2

We're using the IP address now. Basically, the new fix is to change voice.telephony.goog to 64.9.242.172.

Here are the commands for the new fix.

cp /etc/asterisk/pjsip_custom.conf /etc/asterisk/pjsip_custom.conf.old
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 a bunch. That worked for me. However the editor could not find the second file and I looked around in /root/gvsip-naf and there is no such file install-gvsip
 
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Kevman83

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Hello all. I'm a complete newb here when it comes to PBX systems. I have a raspberry pi 3, and I used incredible PBX 13-13 image from a nerd vittles article. I followed the instructions, setup went smoothly. I have 3 Google voice numbers and trunks set up. Have my outbound and inbound routes set up accordingly.

Problem is when I go to make a phone call, there is no audio in either direction from the soft phone or my cellular that I am making the the test call to. Same if I dial my gv number. Also calls in both directions disconnect after 16 seconds precisely.

I am at a loss here as to what I need to do? I read opening ports 5060 and 5061 may help, yet I know that is a security risk. I checked my router and I couldn't find an option to enable or disable SIP ALG. Is there any ports I could open that may help my soft phone work properly?

Edit: Forgot to add that all 3 gv trunks show up as registered following the sed commands on another nerd vittles article.
 

ajonate

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I am at a loss here as to what I need to do? I read opening ports 5060 and 5061 may help, yet I know that is a security risk. I checked my router and I couldn't find an option to enable or disable SIP ALG. Is there any ports I could open that may help my soft phone work properly?

Try opening 10000 thru 20000 and see if that helps. Otherwise you should be looking at your NAT settings.
 

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