Fix the issue: pjsip endpoints go unavailable. freepbx 16/asterisk 13
$30-250 USD
Closed
Posted over 4 years ago
$30-250 USD
Paid on delivery
We have the issue in the production FreePBX 16/asterisk 13.
After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip.
FreePBX is a virtual machine with a public IP (direct). Endpoints are Acrobits sofpthone users (android/iOS) connecting via WAN. there is nothing in between. All end-users use TLS+SRTP. and Acrobits Push. So ping is huge sometimes. I suppose qualify option may be the cause here.
Official FreePBX forum treads ignore the issue and ask to order their paid support.
As the issue is in the production system, there is no place for experiments and we can work only via remote session. I have all the accesses required.
The issue is urgent.
Please ping me for details.
Hi what it is the exact version of Asterisk, this could be an Asterisk bug, if is a bug I just can help you to indentify it and will be needed upgrade of a newer version
привет, не поймали в чем реальная причина, где собственно проблема? Решается ли проблема если установить перед астриском SBC, к примеру?
Если не решите проблему пишите, в течении пары дней смогу посмотреть.
recently i've a done for my customer using pjsip as trunk as well as endpoints and they are pretty stable and i will be happy to help you in this issue and can fix the issue earliest
let me know if you want to start asap
I am currently working on different voip projects both at client end and sip provider end. I have more than 6 years of experience in VOIP solutions and more than 8 years of experience in Linux and Networking.
I got asterisk professional training and certification from authorized Digium Center in Dubai.
Asterisk/Freepbx SIP/IAX/PRI trunks
Asterisk /Freepbx twilio elastic trunk
Asterisk /Freepbx call termination and origination
Asterisk/Freepbx/Elastix/ dynamic ivr using text to speech engine and save DTMF feedback of users in Database..
Asterisk/Freepbx Dialplans
Asterisk/Freepbx IVR , Automatic Call distribution
Asterisk/Freepbx Queues
Asterisk/Freepbx dynamic logins
Asterisk/Freepbx Call Recordings
Asterisk/Freepbx Migration to different cloud technologies.
Customization of asterisk based Voicemail service
Asterisk/Freepbx based cost effective voicemail development using IBM Watson Speech Recognition API
Realtime asterisk deployment
CRM integration with asterisk
Speech recognition API
Text to Speech API
Elastic Server configuration trunks, DID, inbound and outbound routing
Elastix server troublshooting and support
Wazo installation and configuration
Wazo server deployment
Wazo and IBM watson api integration
Wazo endpoint configuration
Wazo Trunk deployment
Wazo and Sip trunk Carriers deployment
Freeswitch Installation and Configuration
Freeswitch/Fusionpbx multitenant deployment for Telecom operators
Freeswitch/Fusion Sip profiles
Freeswitch/Fusion dialplan
Hi
I am systemadministrator. We use asterisk at my work and I am in charge of maintaining it.
Which version of asterisk are you using? It looks like a network problem. What firewall are you using? Is it properly configured?
A huge ping can be a problem sometimes (when > 3000ms).
If your asterisk is on a public network it might get under attack sometimes so it does not help.
Looking forward working with you
Vincent