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SIP to PRI gateway 2
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![[3eyes]](/data/assets/smilies/3eyes.gif "[3eyes] [3eyes]")
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As I stated in my original post. My customer has a MICS. In anticipation of your next question, they are using features such as group listening, etc. that are not available on the UCx. They are completely satisfied with the functionality and reliability of their current system. My customer is just looking to save money through a one-time gateway investment enabling them to use SIP.
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I've had one install now with a Grandstream HT818, as recommended by the fine folks here. SIP incoming, goes into the HT818, this device has 8 ports so you could have up to 8 analog "lines" coming out. The device can let you control hunting as well, since that can't be done on a regular SIP line without a digital PBX.
One thing to note, when you're getting SIP with these devices. Make sure you're getting a SIP trunk, and not some hosted version. I've had this now with two providers who think their hosted system works better with these devices, when in reality I needed a SIP trunk. The hosted systems treat ports on the ATA devices as extensions. So on your MICS you may think of it as Line 1, but the provider see's it as extension 201. Line 2 is ext 202, etc.. This becomes a problem when you want hunting/roll over to work, since the only way we could get it to work was to ring all 3 incoming "lines"(extensions) at a site, at the same time. Then voicemail was an issue as well, since this site did not have an on site voicemail system, they were using the host provided voicemail. So they were not receiving the stutter tone on Line 1/Ext 201, even though the calls were directed to go to Ext 201's voicemail. Long story short, it was a shit show. SIP trunks is what you'll want lol
One thing to note, when you're getting SIP with these devices. Make sure you're getting a SIP trunk, and not some hosted version. I've had this now with two providers who think their hosted system works better with these devices, when in reality I needed a SIP trunk. The hosted systems treat ports on the ATA devices as extensions. So on your MICS you may think of it as Line 1, but the provider see's it as extension 201. Line 2 is ext 202, etc.. This becomes a problem when you want hunting/roll over to work, since the only way we could get it to work was to ring all 3 incoming "lines"(extensions) at a site, at the same time. Then voicemail was an issue as well, since this site did not have an on site voicemail system, they were using the host provided voicemail. So they were not receiving the stutter tone on Line 1/Ext 201, even though the calls were directed to go to Ext 201's voicemail. Long story short, it was a shit show. SIP trunks is what you'll want lol
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