SIP SIMPLE Messages (chan_sip) - Problem with offline endpoints

Hello guys,

I tried to configure SIP SIMPLE messaging on Asterisk 13 using chan_sip according to this guide: SANJAY WILLIE'S Human Language. Asterisk | Nagios | OpenSource | Microsoft | Security: Asterisk 10 or 11 Messaging (SMS/SIP Messaging) with offline message sending and it works fine when both extensions are online.

However, I didn’t succeed getting it to work with queuing of messages - when one of the endpoints is missing. I tried both the author’s script and the one mentioned in comments.

The asterisk console reports that message do get sent to the script for queing, but no .call file gets created and when the receiving endpoint comes back online - it doesn’t receive a message.

Anyone managed to get offline queuing of SIP SIMPLE messages working?

First, chan_sip is history. It will be increasingly difficult to get any help. I think the script will work only for local phones, unless you allow any phone from the WAN side. Not the smartest idea nowadays.

That said, Asterisk knows whether a call file runs successfully or not and you can instruct the call file to repeat its efforts a couple of times. You do not need the device state stuff.

Upgrading the setup will cost a lot of money in developer time and to be honest is not worth it, so I am trying my best to get the most out of it, despite it being ancient. This will not be a system, where anyone will be allowed access - it is just for my friends/known people/relatives, however as I stated in my other thread, it runs a billing solution, which manages Asterisk - and I need that solution as I have to somehow rate outgoing calls that people will make. Otherwise, I would have gone with FreePBX/VitalPBX and pjsip. Unfortunately, the said billing system creates only chan_sip extensions. I have calls working and looking into chat just as an extra functionality as convenience for my users. I have looked into upgrading to a newer asterisk, but I will need to try it on an extra staging machine to ensure everything would run smoothly.

The server is hosted on a Virtual Machine outside of my house. All extensions that I created are able to register. I do get some probes for unknown extensions, but fail2ban seems to block them.

How would I instruct the .call file to repeat its effort? In the script?

First hit googling for ‘asterisk call files’.

Check out ‘MaxRetries’ and ‘RetryTime’.

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