Those iptables are from fail2ban, but do not show any rules to block any traffic on your lo interface.
Just post the lines from two logs from just before a crash to just after.
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URGENT: Asterisk crashing FreePBX?
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Asterisk Manager Password Change
Sorry, I had changed it in the file too, but no workie.
A reboot brought it back.
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Zulu Desktop Screen Sharing
Thanks for the clarification!
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Multiple inbound calls over an hour turns out to be one long call
You might want to try over with your grepping, the bit in the square brackets before the C-00002520 one you used is far more “unique” and might show a little more of the legs concerned.
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Fallover analog and digital line?
But you could construct a trunk between your box and his to use his channels when you don’t but he does.
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URGENT: Asterisk crashing FreePBX?
@dicko There’s nothing about crashing, just the phones registering. 192.168.1.8 is my gateway, and that’s why I’m suspicious. Only when I use iptables -F do the connections return to normal.
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BLF Warm Spare Failover not working
Thank you @lgaetz. I think i have found the problem with your help.
All peers are registred instandly but there are no subscriptions on the backup machine.
Is there a way i can force a new Subscription from the Asterisk side when a Failover Scenario starts? I think the phones will do the subscription again after 1800 Seconds as i can see on the Primary machine, but thats a bit too long for this type of Failover Scenario
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Line 2 works, but line 1 reports: Unable to retrieve PJSIP transport 'udp,tcp,ws,wss'
FreePBX 14 on Centos 7.5
Grandstream GXP2000
I have two inbound routes and two extensions (200, 400) supporting two phone numbers. Both are configured essentially identically, but only one can receive calls. The other (200) fails immediately with
ERROR[14481] res_pjsip.c: Unable to retrieve PJSIP transport ‘udp,tcp,ws,wss’
This is the only log message when the call fails.
I can call out on both lines with no problem – when calling my cell phone, the caller ID on shows the correct originating phone number.
Log of registration messages:
VERBOSE[726] res_pjsip_registrar.c: Added contact ‘sip:200@192.168.71.5:5062’ to AOR ‘200’ with expiration of 3600 seconds
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 200/sip:200@192.168.71.5:5062 has been created
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Endpoint 200 is now Reachable
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 200/sip:200@192.168.71.5:5062 is now Reachable. RTT: 43.316 msec
VERBOSE[726] res_pjsip_registrar.c: Removed contact ‘sip:400@192.168.71.5:5064’ from AOR ‘400’ due to request
VERBOSE[9906] res_pjsip/pjsip_configuration.c: Contact 400/sip:400@192.168.71.5:5064 has been deleted
VERBOSE[726] res_pjsip_registrar.c: Added contact ‘sip:400@192.168.71.5:5064’ to AOR ‘400’ with expiration of 600 seconds
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 400/sip:400@192.168.71.5:5064 has been created
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Endpoint 400 is now Reachable
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 400/sip:400@192.168.71.5:5064 is now Reachable. RTT: 42.140 msec
The pjsip.endpoint.conf:
[400]
type=endpoint
aors=400
auth=400-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=g729,ulaw,alaw,g726,g723,g722,g719
context=from-internal
callerid=Home <400>
dtmf_mode=rfc4733
mailboxes=400@device
mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=no
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en[200]
type=endpoint
aors=200
auth=200-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=g729,ulaw,alaw,g726,g723,g722,g719
context=from-internal
callerid=Office <200>
dtmf_mode=rfc4733
mailboxes=200@device
mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=no
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en[anonymous]
type=endpoint
context=from-sip-external
allow=all
transport=udp,tcp,ws,wss[dpma_endpoint]
type=endpoint
context=dpma-invalid
I have deleted and recreated the OSS Endpoint configuration.
I have deleted and recreated the extension.
Any idea what the problem is/what else to try.?
Thanks.
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URGENT: Asterisk crashing FreePBX?
And there I read
. . .
When I enter the GUI, it displays the message “Can not connect to Asterisk”. When I start the asterisk manually, the GUI stops responding. How can I solve this? PLEASE ITS URGENT!
Edit: After a while, FreePBX starts the Asterisk, and then, GUI crashs again…
. . . .
That was explicitly about crashing and now the thread is about registering ?
Either start a new thread or at least post something substantive. It really will be apparent in the logs, add /var/log/fail2ban.log if you think it relevant . . .
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URGENT: Asterisk crashing FreePBX?
@dicko I think I was wrong. I just wanted to say what was contained in that log. The discussion is still about the crash, and in the logs there is nothing very useful. I think it’s still the iptables thing
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BLF Warm Spare Failover not working
Maybe you could have a trigger that looks at if the trunks are active on the spare. If they are you could trigger a new subscription. You would probably need to add a few more pieces, so it doesn’t keep doing it, but that might work.
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BLF Warm Spare Failover not working
Thank’s that’s what im looking for. How can i trigger a new Subscription to the phones from the Server side via cli?
My only solution i found was setting the session timer in Yealink and Snoms config to a value like 60s.
But if the Server can force a new Subscription via cli i can leave the setting on the phones on 3600s.
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BLF Warm Spare Failover not working
You can’t have the PBX do it. The phone has to intiate it. You would have to get yealink to support it when the phone fails over to subsribe all their hints again.
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BLF Warm Spare Failover not working
The phone does not fail, my problem was the default Setting of 3600s
I tried it with Snom and Yealink:
If the phone just “Re-Registers” they do not Re-Subscribe until the Session-Expire timer ends.
So when i switch to the failover PBX and i wait about 1 Hour then it should work.
I think i will override the Session timers in Yealinks and Snoms Config to 60s
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BLF Warm Spare Failover not working
Well they should handle that better. That’s crappy software design for them not to subsribe to all hints on failover. You are going to overload asterisk if you resubsribe every 60 seconds. Asterisk BLF system is not great to begin with and that will only cause you more issues.
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BLF Warm Spare Failover not working
I just re-read your comments above. The phone is not dual registering but using FQDN. Ya that won’t work very well as the phone doesn’t know the PBX has changed as far as BLFs are aware. How are you going to make sure all phones use a DNS server that will honor such a short TTL like 60 seconds. Lots of DNS servers will ignore such a short TTL
FreePBX HA for instance handles this as it’s realtime sync so all the hints are synced so nothing gets missed so on failover everything keeps working.
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BLF Warm Spare Failover not working
You are absoluteley right 
I have not tried it with Sangoma Phones, Maybe they handle this better. If so i am not aware of using Sangoma S500 Phones instead of Yealink T46S for my next projects.
Thanks for the note that asterisk will probably overload.
I just thinked about leaving this things and just initiate a “check-sync” command with fwconsole epm command on the backup PBX. So phones reboot after they re-registred on the Failover PBX and then everything should work. I think it is okay for a Failover Scenario that i hopefully never will need. This is nothing for a 24/7 customer requirement but i dont accept that customers at the moment as i do not have ressources for this at the moment.
I only sell Cloud Hosted PBX Systems with a own Dedicated DSL Line provided from our partner. So i can make sure everything works (Short Traceroutes, Perfect Ping and the TTL60 works.). If the customer uses his own Internet Access then it is on his own risk.
I dont know if HA already supports multiple Server Destinations without own internal network. (Each Server only has a Public IP)
The normal customer with 30 Extension is not willing to pay the high price for HA. I know it is worth it, but so the monthly fee for the customer is about 6-10 Times higher than without HA.
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Multiple inbound calls over an hour turns out to be one long call
Thank you, but grepping by that preceding number does not show up anything new.
In the timeframe of the call, there were only three different numbers shown before the “[C-00002520]”:
- 3486 - 3 three lines like
[2018-07-05 08:23:06] VERBOSE[3486][C-00002520] netsock2.c: Using SIP RTP TOS bits 184
- 44536 - 3,029 lines like:
[2018-07-05 08:23:06] VERBOSE[44536][C-00002520] pbx.c: Executing [4809616003@from-trunk-sip-Digium-SIPTrunk:1] Set("SIP/Digium-SIPTrunk-00000b71", "GROUP()=OUT_13") in new stack
- 52554 - two lines like:
[2018-07-05 09:20:12] VERBOSE[52554][C-00002520] bridge_channel.c: Channel SIP/1015-00000b98 joined 'simple_bridge' basic-bridge <8a4c9a8e-ca50-429e-a9a2-e72e22b45f11>
The total was 3,034 lines - the same as when I grep with just the “[C-00002520]”
Any other thoughts?
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User Control Panel Call History Halted
I have a FreePBX distro that the call history in UCP for all user’s extensions are showing calls from August 29th, 2017 and older. Nothing new.
I have completely removed the UCP module, and reinstalled it. All other modules and RPMs are up-to-date. PBX Version 14.0.3.6 and System: 12.7.5-1805-2.sng7
Any ideas on how to fix?
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Can I change the type of transport?
Anyone?
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