I turned on the firewall, but if I turn it on, do the extensions configured with the zoiper work even without adding exceptions to the list of ip firewalls?
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Pbx in standby?
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HT813 on FreePBX 14 trunk hangs up at 30 seconds
Greetings, I’ve got a new FreePBX 14 installation with a Grandstream HT813 FXS/FXO SIP Server connected to a POTS line. Incoming and outgoing calls are working, but they hangup after 30 seconds. This appears to be the failing segment of the log. 172.31.5.164 is the HT813 SIP Server. The FreePBX server is 172.31.4.3 on a /22 (255.255.252.0) network.
Can anyone shed any light on this subject as to why the “RTCP from 172.31.5.164:5015: Failed first packet validity check” is happening?
Thanks, Mark
[2019-10-31 22:04:21] DEBUG[3421] res_pjsip/pjsip_distributor.c: Searching for serializer associated with dialog dlg0x7fb3741213b8 for Response msg 200/BYE/cseq=21026 (rdata0x7fb3ac12a9c8)
[2019-10-31 22:04:21] DEBUG[3421] res_pjsip/pjsip_distributor.c: Found serializer pjsip/distributor-0000003b associated with dialog dlg0x7fb3741213b8
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Source of transaction state change is RX_MSG
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Received response
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Response is 200 OK
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Received response
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Response is 200 OK
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: BYE received final response code 200
[2019-10-31 22:04:21] DEBUG[28227][C-0000005c] res_rtp_asterisk.c: Got RTCP report of 12 bytes from 172.31.5.164:5015
[2019-10-31 22:04:21] DEBUG[28227][C-0000005c] res_rtp_asterisk.c: 0x7fb3742a2de0 – RTCP from 172.31.5.164:5015: Failed first packet validity check
[2019-10-31 22:04:21] DEBUG[19332] manager.c: Examining AMI event:
Event: HangupRequest
Privilege: call,all
Channel: PJSIP/6000-0000015a
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 14055551212
CallerIDName: NAME HERE
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: app-pbdirectory
Exten: pbdirectory
Priority: 4
Uniqueid: 9999999999.999
Linkedid: 9999999999.999
Cause: 18
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HT813 on FreePBX 14 trunk hangs up at 30 seconds
The RTCP error you noted is unrelated to the call drop – the first line of log posted indicates that Asterisk has already received a response to BYE (that it presumably previously sent as a result of detecting some other problem).
The usual cause of this issue is that the ACK for the 200 OK response to INVITE is not properly sent or received. In the HT813, confirm that STUN Server (advanced settings) is blank and NAT Traversal (FXO port) is No.
Also, confirm that in Asterisk SIP Settings, External Address and Local Networks are properly set, and that you restarted (not just reloaded) Asterisk after making any changes.
The basic IP settings for both devices should have the correct subnet mask, so that packets sent between them do not pass through any router or firewall.
If those are all correct (or changing them didn’t help), at the Asterisk console, type
pjsip set logger on
make a failing call and post the Asterisk log, which will now contain the SIP conversation.
For a long log such as this, paste it at https://pastebin.freepbx.org and post the link here.
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How to read a Zulu changelog
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FreePBX 15 is Released!
Thanks for catching that @jfinstrom ! I thought we’d already had it pushed out but maybe it hadn’t made it yet. Also, thanks for your contributions as well on the FreePBX 15 release.
Matthew Fredrickson
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HT813 on FreePBX 14 trunk hangs up at 30 seconds
Oh My Goodness…
Many thanks, Stewart, I checked all of the network settings on the HT813 and the server, and the SIP Settings. They all looked great.
I restarted the machine (as you suggested), and the calls started completed correctly.
Apparently I had had entered everything, but didn’t get a restart…
Many thanks for pointing out the obvious, as that’s what I missed.
Mark
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Clicking on apply causes pjsip endpoints to drop
Hi Thank you, i’m on 15.0.6.7 so i will try this.
I think it could be something else even with “allow transport reloads” set to on I have not seen it before where applying settings will cause all endpoints to unregister like this and they will not come back online without restarting asterisk
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SMS Not Working in UC or ZULU. What do I need to do?
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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Purchasing Commercial Modules - Instance License or Customer License?
Thank you for your testing and reply! In short of forking out $1500 x2, HA/failover is not as easy as it needs to be. We should be allowed to unlimited detach and reattach via some sort of Sangoma Portal/ licensing API, we own the license at the end of the day.
Luckily GCP seems very stable and we’ll never need this knock wood
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LLDP configuration
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UCP Original mailbox filed no more shown. voicemail
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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After updating FreePBX there were warnings in the console at outgoing calls
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Purchasing Commercial Modules - Instance License or Customer License?
And unless you’re running FreePBX v13 there is no current HA support in v14 or v15.
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How to handle missing calls
Hello! Newbie here. It is my first time setting up a Freepbx server so i have a few questions.
I have setted up a trunk sip from my provider, created an extension, then an inbound route that leads to the extension and an outbound route for dialling out.
I have only 1 number from my provider and at the moment i don’t need any other internal phones.
I am using a softphone application (zoiper) at my smartphone in order to receive and dial phone calls. While I am outside my office, i am using VPN in order to still receive phone calls.
When my GSM signal is low or my phone runs out of battery, is there any way to track down missed calls while my softphone app is unregistered? How do you recomend me to handle the missing calls when there is no end point(softphone or any IP phone) connected at the freepbx server?
thanks!
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Conference number problem
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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How to handle missing calls
If the phone is unavailable (for any reason), the call should go to the extension’s voicemail, which is maintained on the server. Zoiper allows for VM notification, so you should have plenty of notice.
If you need more immediate notification, you can combine the Voicemail system’s “Pager” option to send an SMS to your phone using your provider’s Email-to-SMS gateway. This way, you get an immediate text message when the phone gets back into GSM range.
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How to handle missing calls
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Cambio de CallerID a extensiones internas de Asterisk hacia PSTN
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Yum upgrade or yum update?
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Grandstream gxp 2140 MWI inconsistent messages waiting
We use Grandstream phones with FreePBX because they’re much more affordable than others. But we have erratic behavior with them.
This time … the MWI (message waiting indicator) lamp sometimes blinks after a message is left. Other times it waits several days before blinking.
I’ve tried adjusting the Extensions > someextnum > Advanced > MWI Subscription Type …
with Auto - it sometimes works properly
with Solicited - it sometimes works properly
Can anyone suggest other settings to check - either in FreePBX or on the Grandstream phone?
Thanks in advance, Jason.
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