jlr
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Posts posted by jlr
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DTMF - it is hit or miss as to when it works making it impossible to navagate phone menu's when calling Credit Card Co, even My sip provider SOTEL - I have had the problems with aastra 6757i, snom 820 & a pap2 adapters. Any help would be great - I like the PBXnSIP system.
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.
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I want to do exactly this.
I am being told I need to setup this at the SIP trunk level.
That doesn't sound right to me...Do I need to set this up on pbxnsip AND the sip trunk vendor?
Seems this should be simple but i'm confused now,
Matt
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.
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on a Cisco 3550, when i first plug it in it powers up but takes a while for the phone to actually connect/register its extension, it starts the countdown process repeating prov.prequest from 20 down to 0, and at about 9th attempt it will register only sometimes, other times it will keep power cycling
with polycom phones i dont have the problem, only with all snom phones i have tried
its something about when the phone drops the ethernet after receiving power
Very strange. You tried using the portfast feature on the switch? Running the most current firmware on the phone?
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Does anyone know what setting the Snom phones are looking for when powered up using POE Switch ?
I'm currently using the value of= power inline delay shutdown 20 initial 300 on my cisco 3550 switch but it seems as if the phone doesn't work correctly
what I'm looking for is how long after it gets power is it looking to drop the Ethernet
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.
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so you believe you had the issues with the SIP Provider? and its not related to the PBX?
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.
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There are many buggy RFC2833 (now RFC4733) implementations. I assume that when you look into the wireshark either the two "2" are glued together or there is no DTMF at all. Sometimes providers signal they they would support out of band in the SDP, then when the media arrives it is inband.
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.
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All these digits are consumed by the PBX to make/take decisions before the phone starts ringing. So, the answer is no.
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?
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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.
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I'd be curious to know as well.
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Those are the only 2 options that you have on the conference.
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!
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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.
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.
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That looks pretty good.
What you can try is to set "Trunk requires out of band-DTMF tones", though I would give that only a small chance. The other thing that you can try is to put the handset into inband mode, so that all DTMF is inband by nature. Higher chance, but this will make it difficult to have DTMF on the PBX.
Of course, it would be interesting what the trunk provider returns in the 200 Ok. My guess is that they return support for telephone-event, but when sending the RFC4733 tones it fails.
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.
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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?
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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
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Is there a way to have unknown calls at least go to a recording of some sort instead of a congestion message?
Example: On Avaya systems if I have an unknown call come in on a PRI I can have it either send a re-order or go to a system intercept/recording.
I'd like to be able to do the same thing on SIP trunks for numbers that may not necessarily be assigned but valid in a range of DID numbers. "You've reached a disconnected or out of service number at xxxx..........".
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Is there a FreeBSD Beta build floating around that's available yet?
Thanks!
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Are there any known issues with the FreeBSD build of this update? I've downloaded the new 3.4.0.3201 build and received an Exec format error and that the binary file isn't executable.
Currently running 3.3.1.3177 on release 7.2 of FreeBSD.
Call Parking w/snom & v4.0.1
in Call Treatment
Posted
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?