jlr

I'm trying to get call park operating on a snom 820 phone and version 4.0.1 of the software. I've tried to do a basic call park by placing the call on an exclusive hold and dialing *85. When I dial the *85 I get a busy signal and when I pickup the original call I can still here the MoH on the other end of the call plus the audio from the handset. The docs on the site aren't very helpful on this topic. I have multiple registrations of the number in question and would just like to be able to place it on a park if answered in one place so I can pick it up in another. Any ideas?

There are some options as to whether to send DTMF in-band or out-of-band in the trunk settings. I experienced similar issues both ways and ended up opening a ticket with the SIP trunking provider. The ITSP ended up moving my trunk to another one of their servers which corrected the issue. Check with your provider and see what they can come up with.

You should only have to add the DID number in the account settings for the domain (eg. the Auto-attendant, and extension accounts). The screenshot in the other post gives you an example. You shouldn't have to do anything with the trunk itself.

Very strange. You tried using the portfast feature on the switch? Running the most current firmware on the phone?

What symptom are you seeing specifically? You shouldn't have to do any real tweaks to PoE on the switch. I've run a snom 820 on Cisco PoE gear without making any changes.

It's certainly a possibility. I kept looking at everything on the PBX and handset side and once I verified it was and looked ok put in a ticket with the carrier who assisted in the troubleshooting process and found the problem to be on the trunking server I was connecting to (though it had been working for some time before that). They made some changes apparently, I was moved to another primary and secondary and the problem went away and has been fine since. I only had the issue with one of the trunking providers. Do you have multiple providers that this is a problem with? Worth a try asking the provider.

I had the same issue where the SIP trunking provider changed me over to another SIP registration proxy due to DTMF issues (different implementation apparently). Once on the newer proxy the problem went away. Sounds very similar to what I had issues with.

Wouldn't this be 3rd party/SOAP app stuff? Examples being a banking client that takes in a account number that needs to be passed to the agent/PAC from DTMF inputted by the caller?

Out of curiosity, is this coming in from a SIP provider? I recently went through a DTMF issue on the outgoing side that turned out to be the provider not coding it properly.

I'd be curious to know as well.

Would it be possible to add features for moderators from the current limited abilities? Features such as: Record conference Lecture Mode (mutes all but the moderator(s)) Participant count Dial-out to add other participants Just curious of quick items that could be added. Thanks!

Seems to be something with Vitelity's SIP trunking service. I moved my trunk preference up for outbound to Voicepulse and DTMF seems fine through calls there. At least it's been narrowed down. Time to go to them and compare notes.

I tried setting the trunk to requiring in-band. While it's interesting to note that some IVRs seem to get the tones many others don't. I've tried a few different Avaya based along with some unknowns and sometimes it works and others it doesn't. Almost like the DTMF generated is "close enough" for some but not others.

They (Vitelity) do support RFC2833. In the trunk set-up I have "Trunk requires out of band DTMF tones" set to no. Here's a glimpse of what I see in the log: [9] 2009/11/09 16:11:07: Message repetition, packet dropped [7] 2009/11/09 16:11:09: Received RFC4733 DTMF on codec 101 [8] 2009/11/09 16:11:19: Packet authenticated by transport layer [7] 2009/11/09 16:11:19: c26fe9c8@pbx#2101923944: Media-aware pass-through mode [7] 2009/11/09 16:11:19: Other Ports: 1 [7] 2009/11/09 16:11:19: Call Port: c26fe9c8@pbx#2101923944 On the Snom 820 the DTMP Payload type is grayed out I believe from the auto-provisioning pbxnsip does for Snom's. RTP/RTCP: Dynamic RTP port start: 49152 Dynamic RTP port stop: 65534 DTMF Payload Type: 101 RTCP Support: . on off RTP Keepalive: . on off For the SIP Identity Settings the 'DTMF via SIP INFO:' is also grayed out to off from auto-provision.. Ideas?

Using various VoIP handsets on 3.4.0.3201 including a snom 820 I receive mixed results when using defaults for DTMF with my Vitelity SIP trunk. On very few occasions the DTMF works but usually doesn't when entering digits for AVR type systems. Any ideas? TIA