hosted

so we have to be logged into the user portal? I was thinking it would be nice if I hung up the bad caller and did *92 and it blacklisted the last call kind of thing.

yea I can send a log. its a little hard to explain. CompanyA: let say extension 100 has DID 8012345678 assigned to it and extension 101 has DID 8012345600 assigned to it Now a third PBX who spoofs their ANI (CompanyB) makes outbound calls for this companyA (via PSTN) uses 8012345600 as their ANI. This companyB can not call any DID whatsoever (to CompanyA) because their ANI matches a DID in the main companyA. This is extremely rare i know. But to me if companyB ANI matches inbound DID to companyA why would you 404 the call?

Thanks pbx support. The cases we run into are people calling an IVR and pressing 1 to transfer to an external DID (answering service for example). they want that call recorded to monitor the answering service. Also when a call dual rings a cell phone. that external call needs to record as well. I can see the possible legal aspects, but that should be left to the end user. a simple "your call maybe recorded" greeting normally overrides that legal aspect. PS, I also am in a little hot water over this as I was charging someone for full time recording and well... yea its an IVR system and all their calls go external.

PAP2 can do T38 now? hmm i though you had to use the 2102.

the black list works for me when I manually put them in the address book.. but how does it work when you on a call? If I press *92 after the call it says feature not available. if I put the caller on hold and *92 same thing. what am i doing wrong?

Yes this still does not work. DID calls an IVR you press 1 which calls an external number (example 8012345678) when the external call answers the call is NOT recorded. This is frustrating.

what would be cool is if you could have a listen only code. then you dont have to mix to many conversations and get a lot more people to listen. which is what you want for large calls anyway.

were still waiting on the ability to transfer a call back to the PBX when forwarded to a cell phone.. This would be such a great feature I know its been talked about before. For example a call if forwarded to my cell phone from snomone I can press ## and get PBX dial tone to transfer a call back to a receptionist or something. need

we can resell hosted licenses as well. Call or email me if you have questions. David Burr burr@nexsip.com 801-320-1000

if you have 2 DID's (for example 8012345678 and 8012345679) and you call into the PBX and you ANI is one of the above DID's the system will send back a 404. example you call 8012345678 with ANI 8012345679. I think this is a bug. db

bump

snom/pbxnsip now have 2 built in certs. can I replace them with my own? or are they specific for snom phone usage or what... think the snom phones ignore the auth and still SSL/TLS anyway..

on your 3.0GHZ servers we try to keep the concurrent calls to about 30. I have seen them go a little higher, but that is our magic number. All of our hosted customers are on dedicated blade servers we do not share resources. our blades go from 10 concurrent calls to 35. our 1.5Ghz blades are 10 concurrent calls our 3.8GHZ blades are 35. db

The signaling might be TCP/TLS however the T38 transport will still be UDP, and still subject to the same packet loss issues. db

We have a customer using our https fax service today over a clear wimax modem. works great, no way T38 would survive that. As far as using a virtual host on 150 users.. You would pretty much need a dedicated server to do that so it would be wiser to rent a server instead of a VM. For pricing information on the HTTPS FAX ATA's email me burr@nexsip.com db

So a call coming in on a trunk going to an IVR and dialing out a trunk DID *no extension involved* will now record with 3981?

make sure your polycom_sip.cfg says tcpIpApp.sntp.daylightSavings.start.date="{tz dst-start-day}" for some reason the my 3.4 was wrong.

So what if we dont use an extension? DID to an AA then it dials a trunk..

your pcap only show traffic from 192.168.19.57 to 192.168.19.101 so we cant see the entire SIP messaging. so a little hard to diagnose.

which is why we limit it at the trunk level

I do this by limiting CO LINES in the trunk settings.

We have the same scenerio. If a call comes in to an IVR and press 1 to redirect to a cell phone. that call is not recorded. did this work in older versions? We are charging a customer for recording so I need to figure out how to make this work.

PBXNSIP 4.2.0.3981 Linux 32 bit running on centos 5 64bit I had 2 servers crash with this version on me TODAY. not stable.

we dont like to go above 40 calls. someday when pbxnsip/snom gets multi-RTP cores rolling it will be a good day.

This is easy to troubleshoot. load wireshard on the windows box capture the RTP and it will tell you exactly what devices are causing issues. If you find ALL devices are lossing packets you have 2 places to look your switch or your server. Most likely your server. I would not run VM. you might have "juice" but you will probably have "jitter" which is much worse