catejust

Thank You!

Hello! I couldn't find any documentation for any of the following parameters found in pbx.xml and was curious exactly what they control / why they are there? <trunk_reinvite>false</trunk_reinvite> Trunk reinvite -- Is this whether trunks will accept reinvites? <allow_pass_through>true</allow_pass_through> Allow Passthrough -- I know that allowing passthrough just means that the PBX is proxying RTP.. Would disabling this force peer-to-peer invites to be sent out? <save_registration>false</save_registration> Save registration -- Will this remember a registration and prevent 410 Gone responses when an endpoint tries to re-register with the same sequence number? <jitter_advance>true</jitter_advance> Jitter advance -- Is this a jitter buffer? Can it be adjusted at all? Thank You!

Is that ring_duration setting not exclusive to the M9?

Hello! Does anyone have any experience on how to make snom 7xx (720 and 760) series phones stop ringing when they receive an invite and then never receive a SIP CANCEL (for example if the phones lose registration). We are tesitng some failover techniques and if the phones have already began to ring after they receive a SIP INVITE, we need to be able to define, hopefully per identity, how long they will ring without receiving a CANCEL. Thanks!

Good Day! I'm curious about the frequency probability of EPID generation across several PBXs. If we were to have many unique instances of snom ONE running as their own, independent call server, how likely is it for an EPID to repeat? I know that statistically speaking, there is a one in ten billion chance of an EPID repeating. Having said that, since there is not a truly good way of generating random number, my question references more specifically the algorithm snom ONE is using. If we had ten instances of snom ONE running across ten separate servers, let's say all of these snom ONE instances send calls to an e911 provider through a single SIP gateway. When the e911 provider called back into our SIP gateway and we search all ten servers for the EPID, how likely is it that the EPID would be a duplicate and the call would route to the wrong endpoint? Thank You!

Thank you gotvoip! I appreciate you sharing your knowledge very much.

What do these two settings do? Unless I don't have up-to-date documentation, I can't seem to find any information about what these two settings do?

We're having an issue during calls. The issue happens when we make an outbound call from any of our registered endpoints (Polycom/Panasonic/Snom) out our PSTN trunk, and we reach a voicemail system on the PSTN. After the "BEEP" tone that voicemail systems use to indicate when to begin recording a voicemail, there is obviously no sound being transmitted from the far-end voicemail system to Snom ONE. During this time that we are recording a voicemail, there is an extremely loud static/gargling sound heard on our endpoint. This ends as soon as audio begins to be transmitted again from the far-end. So, we heard this static sound ONLY when Snom ONE is not receiving RTP from the far-end. How can we disable this behavior?? I had opened a pbxnsip TT on this quite a while back and never got a definitive reply.

Has anyone experienced this? All calls that come into to certain agent groups show up without a duration both int he nightly email and also in the agent group call log. There is no Waiting Time, Ring Time, Talk Time, or (Hold Time).

Hello! I wanted to see specifically what the option RFC3325 for "Remote Party/Privacy Indication" is? We are currently using RFC3325, but don't hide as our Remote Party/Privacy Indication because it is the only way for the caller ID of inbound PSTN calls routed out to cellphone numbers to display correctly. If we use RFC3325 P-Preferred, outbound calls to certain cellphone numbers display as "Unknown" and our switch confirms this (the switch actually shows the pbx is sending "pbx0X") as the ANI instead of an actual phone number. RFC3325 but don't hide is working for ANIs, but it does not let caller ID blocking work. (i.e. *67 does NOT block outbound caller ID). So, I'm hoping this RFC3325 custom option might be the answer to our issues but I have yet to find any documentation on it. Could you please provide what this does specifically? Thanks!

Hello! I was just curious if anyone knew whether or not at any point Snom One would begin featuring call-routing rules based on the caller ID of a party calling into the pbx? Thanks =)

What if this issue is happening a few hundred times a second? The logs are showing it literally with that frequency.. If a pbx were on a lan.. would there be anything that might cause this?