And because of all of that, you lose flexibility and are locked in to their service.
Running on the assumption that there are always snowflake scenarios in all areas of business, and ignoring them unless you are one, there are zero reasons to ever buy your phone service from the same company providing your internet service.
Don’t disagree.
Well I wouldn’t ebay equipment… I would get proper warrantied gear.
But yes, your proposed solution is perfectly viable. it is how a lot of hotels work.
OK. Thanks.
So if I buy SIP service from CenturyLink then I DO need a SBC to connect their SIP service to my FreePBX! Thanks for the explanation.
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We ported in a number for a client with a NEC DSX 80, they have it setup with an auto attendant, calls get delivered fine from our cloud freepbx to the on-premise grandstream ht818 ata, but when auto attendant options are pressed, nothing occurrs.
I have checked the following:…the extension on freepbx has dtmf signaling set to RFC2833
the ht818 profile 1: dtmf payload type: 101, preferred dtmf method: rfc2833
Not sure if there’s anything else I should check. Any ideas?
This seems unusual. Most telcos require you to use their SBC. The SBC lease is included as part of your contract. Their tech installs the equipment at your location and helps you connect it to your PBX. Depending on the details, you may need a separate NIC or VLAN on the PBX.
Stewart, yes, in fact, I use Centrylink at one of my customers and this is the norm, they provide a router and SBC, you connect your PBX to their SB…the SBC is the demarc
Enable dtmf debug on Asterisk so you can see in the log if the digits are reaching Asterisk or not.
Not disagreeing with anything said so far, but another perspective is the cost model.
What is the monthly on their new PRI, the rent on the SBC, taxes, fees, 911 interconnect fees, etc.?
What is the monthly on just getting “generic Internet” from them and you choosing one or more competitive carriers (not CLECs, just other ITSPs)? Most ITSPs require a commitment in terms of rates and minimums, plus there are charges “per phone number” and “per minute, per call”.
In my dealings with CLink, the PRI is the differentiator. The cost for delivering your phone numbers over 1.5Mbps PRI are way higher than just getting 2Mbps of pure internet over the same fiber link.
Whichever you choose, you are going to end up in reasonably good shape. Follow the money and you will get to your answer.
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It’s possible that the Grandstream is capturing and interpreting the digit presses for you, which isn’t what you want. Turning on the DTMF debugging will tell you if the digits are even making it to the PBX. If they are not, either the ATA or the DSX is snagging the audio.
I want the DSX to snag the audio, but that’s not what’s happening, but that’s an interesting idea, maybe I’ll try turning off DTMF on the Grandstream.
I already tried rerunning/restarting firewall process.
It looks like the pbx is getting them: I pressed 2, 3, 3. I see those characters in the log. Perhaps the Grandstream is taking those presses and not passing them thru to the NEC DSX?
Here’s a copy of the log:
[2019-05-28 16:32:24] Asterisk 13.23.1 built by mockbuild @ jenkins2.schmoozecom.net on a x86_64 running Linux on 2018-10-03 21:46:52 UTC
[2019-05-28 16:32:38] DTMF[12483][C-0000a393] channel.c: DTMF begin ‘9’ received on SIP/1168-00016e1e
[2019-05-28 16:32:38] DTMF[12483][C-0000a393] channel.c: DTMF begin passthrough ‘9’ on SIP/1168-00016e1e
[2019-05-28 16:32:38] DTMF[12483][C-0000a393] channel.c: DTMF end ‘9’ received on SIP/1168-00016e1e, duration 160 ms
[2019-05-28 16:32:38] DTMF[12483][C-0000a393] channel.c: DTMF end accepted with begin ‘9’ on SIP/1168-00016e1e
[2019-05-28 16:32:38] DTMF[12483][C-0000a393] channel.c: DTMF end passthrough ‘9’ on SIP/1168-00016e1e
[2019-05-28 16:33:33] DTMF[13757][C-0000a3b3] channel.c: DTMF begin ‘2’ received on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:33] DTMF[13757][C-0000a3b3] channel.c: DTMF begin passthrough ‘2’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:33] DTMF[13757][C-0000a3b3] channel.c: DTMF end ‘2’ received on SIP/my-sip-provider–ob-trunk-1-00016e70, duration 395 ms
[2019-05-28 16:33:33] DTMF[13757][C-0000a3b3] channel.c: DTMF end accepted with begin ‘2’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:33] DTMF[13757][C-0000a3b3] channel.c: DTMF end passthrough ‘2’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:33] DTMF[13757][C-0000a3b3] channel.c: DTMF begin ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:33] DTMF[13757][C-0000a3b3] channel.c: DTMF begin passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF end ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e70, duration 395 ms
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF end accepted with begin ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF end passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF begin ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF begin passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF end ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e70, duration 415 ms
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF end accepted with begin ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:34] DTMF[13757][C-0000a3b3] channel.c: DTMF end passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e70
[2019-05-28 16:33:52] DTMF[13791][C-0000a3b4] channel.c: DTMF begin ‘0’ received on SIP/my-sip-provider–ob-trunk-1-00016e72
[2019-05-28 16:33:52] DTMF[13791][C-0000a3b4] channel.c: DTMF begin passthrough ‘0’ on SIP/my-sip-provider–ob-trunk-1-00016e72
[2019-05-28 16:33:52] DTMF[13791][C-0000a3b4] channel.c: DTMF end ‘0’ received on SIP/my-sip-provider–ob-trunk-1-00016e72, duration 75 ms
[2019-05-28 16:33:52] DTMF[13791][C-0000a3b4] channel.c: DTMF end accepted with begin ‘0’ on SIP/my-sip-provider–ob-trunk-1-00016e72
[2019-05-28 16:33:52] DTMF[13791][C-0000a3b4] channel.c: DTMF end ‘0’ detected to have actual duration 60 on the wire, emulation will be triggered on SIP/my-sip-provider–ob-trunk-1-00016e72
[2019-05-28 16:33:52] DTMF[13791][C-0000a3b4] channel.c: DTMF end ‘0’ has duration 60 but want minimum 80, emulating on SIP/my-sip-provider–ob-trunk-1-00016e72
[2019-05-28 16:33:52] DTMF[13791][C-0000a3b4] channel.c: DTMF end emulation of ‘0’ queued on SIP/my-sip-provider–ob-trunk-1-00016e72
[2019-05-28 16:33:55] DTMF[13858][C-0000a3b6] channel.c: DTMF begin ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e75
[2019-05-28 16:33:55] DTMF[13858][C-0000a3b6] channel.c: DTMF begin passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e75
[2019-05-28 16:33:55] DTMF[13858][C-0000a3b6] channel.c: DTMF end ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e75, duration 275 ms
[2019-05-28 16:33:55] DTMF[13858][C-0000a3b6] channel.c: DTMF end accepted with begin ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e75
[2019-05-28 16:33:55] DTMF[13858][C-0000a3b6] channel.c: DTMF end passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e75
[2019-05-28 16:34:02] DTMF[13858][C-0000a3b6] channel.c: DTMF begin ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e75
[2019-05-28 16:34:02] DTMF[13858][C-0000a3b6] channel.c: DTMF begin passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e75
[2019-05-28 16:34:02] DTMF[13858][C-0000a3b6] channel.c: DTMF end ‘3’ received on SIP/my-sip-provider–ob-trunk-1-00016e75, duration 275 ms
[2019-05-28 16:34:02] DTMF[13858][C-0000a3b6] channel.c: DTMF end accepted with begin ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e75
[2019-05-28 16:34:02] DTMF[13858][C-0000a3b6] channel.c: DTMF end passthrough ‘3’ on SIP/my-sip-provider–ob-trunk-1-00016e75
It all depends on the NEC PBX. Certainly doable with Asterisk, but you will have to check what specific SIP capabilities does the NEC have. You will probably need a license for SIP trunking on the NEC if you don’t have one already. Are you trying to keep the NEC for a particular reason? Why not go all in with Asterisk?
Hello all, when I use asterisk -r to get to asterisk console I have a lot of these scolling past when there are calls in progress.
Sent RTP packet to xxx.xx.2.41:34024 (type 00, seq 026108, ts 328936, len 000160)
Got RTP packet from xxx.xx.2.41:34024 (type 00, seq 002040, ts 328880, len 000160)
Sent RTP packet to xxx.xx.2.41:34024 (type 00, seq 026109, ts 329096, len 000160)
Got RTP packet from xxx.xx.2.41:34024 (type 00, seq 002041, ts 329040, len 000160)
Sent RTP packet to xxx.xx.2.41:34024 (type 00, seq 026110, ts 329256, len 000160)
Got RTP packet from xxx.xx.2.41:34024 (type 00, seq 002042, ts 329200, len 000160)
Is there a way to not have these? I have tried core set verbose 0, I have looked under advanced settings, turned off all logging to the console file in log management. These scroll past at such a rate the trying to watch call flow is near impossible.
Thanks,
John
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Hello
I will try to explain
Incoming call to my FreePBX:
CID: + country code - area code - number
DID: + country code - my extension
An outgoing connection from my FreePBX
CID: my extension
DID: area code - number
from asterisk cli
rtp set debug off
(I guess you turned it on at some time but forgot to turn it off 
)
after you do that you will need to
core set verbose 3
to watch any “call flow”