I've just configured a SIP ITSP on my working lab system back in England. I am currently working in Australia so have signed up with an Australian SIP provider and obtained an Australian DID number. This now allows me to make local calls back to my family home in England and also allows them to call me or another family member here in Australia.
I post this here for a findings discussion, as I did some extensive searches and found differing results. I'm also studying CVOICE which covers some if not all of my tasks involved in the setup.
I found that when I enabled the SIP registrar stuff under "sip-ua" that the system tried to register all of my extensions and dial-peers. I resolved this by telling the system not to register using "no sip-register" and "number xxxx no reg" commands under the dial-peers and ephone-dn's respectively.
I then found that there was no registration messages being sent, so to register with the ITSP I needed to create a ephone-dn with my SIP username. The system now registered the SIP account. Is this the only way / proper way of registering and creating the connection to the ITSP?
I also found that calls must originate from my SIP username and DID calls are sent to my username. Meaning, when I made outbound calls, the ITSP didn't like seeing my extensions, so I transformed them to the username. Also, for inbound calls, I transformed my username to the real DID which is configured on a hunt pilot. I havent completed any packet sniffing to determine where in the invite message that my DID number is listed, if at all.
I'm also finding that the 3600 second timeout value seems a bit long. I haven't looked very deep into this but I find that if I leave the system for some time I cannot make or receive SIP calls. I lowered the timeout value to the minimum 60 seconds with a refresh of 50 and now all works flawlessly and stable every time. Inbound and outbound calls work every time this way.
What are other peoples findings on the above?
The ITSP is Asterisk based, apparently. Not sure if this makes any difference.
I'm not able to access my old voice mail messages all of a sudden. The recording says something like 'the message is currently not available'. This has never happened before in all the years I have been using this system. I have t...
If you have 2 ISR routers, one acting as Failover, do we need to have both the same number of SRST licenses on the 2 routers?
No. You will only need the SRST licenses on the primary router. Because this feature...