Toll Free not working is it me or provider

1

I have 5 numbers all with the same provider Twilio

I have 5 inbound routes all identical in nature with the exception of the DID corresponding to each of the 5 from Twilio.

The toll Free number gets a message that it is disconnected (Allison Smith Voice) the other 4 numbers ring thru just fine, we can make and receive calls over the trunk without any hiccup.

Asterisk shows this:
Connected to Asterisk 14.7.5 currently running on freepbx (pid = 2313)
== Setting global variable ‘SIPDOMAIN’ to ‘96...***’
– Executing [+18005551212@from-sip-external:1] NoOp(“PJSIP/anonymous-00004bfc”, “Received incoming SIP connection from unknown peer to +18005551212”) in new stack
– Executing [+18005551212@from-sip-external:2] Set(“PJSIP/anonymous-00004bfc”, “DID=+18005551212”) in new stack
– Executing [+18005551212@from-sip-external:3] Goto(“PJSIP/anonymous-00004bfc”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“PJSIP/anonymous-00004bfc”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“PJSIP/anonymous-00004bfc”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“PJSIP/anonymous-00004bfc”, “1?noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“PJSIP/anonymous-00004bfc”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2018-10-25 08:51:11.398 PDT.
[2018-10-25 08:50:56] WARNING[21865][C-00001663]: func_channel.c:470 func_channel_read: Unknown or unavailable item requested: ‘recvip’
– Executing [s@from-sip-external:6] Log(“PJSIP/anonymous-00004bfc”, "WARNING,"Rejecting unknown SIP connection from “”) in new stack
[2018-10-25 08:50:56] WARNING[21865][C-00001663]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
– Executing [s@from-sip-external:7] Answer(“PJSIP/anonymous-00004bfc”, “”) in new stack
> 0x7feaecde7190 – Strict RTP learning after remote address set to: 54.244.51.28:14428
> 0x7feaecde7190 – Strict RTP switching to RTP target address 54.244.51.28:14428 as source
– Executing [s@from-sip-external:8] Wait(“PJSIP/anonymous-00004bfc”, “2”) in new stack
> 0x7feaecde7190 – Strict RTP learning complete - Locking on source address 54.244.51.28:14428
– Executing [s@from-sip-external:9] Playback(“PJSIP/anonymous-00004bfc”, “ss-noservice”) in new stack
– <PJSIP/anonymous-00004bfc> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [h@from-sip-external:1] Hangup(“PJSIP/anonymous-00004bfc”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00004bfc’

So with all of that I am at a loss, I have tried everything I know to fix the issue. Our toll free gets used a lot and the previous IT has taken all the steps necessary to open the firewall to Twilio’s IPs and Ports as I have double verified them. And I am leaning towards this being my problem and theirs but I need to know for sure.

12.7.5-1807-1.sng7
FreePBX 14.0.3.19
Asterisk 14.7.5

2

looks like you’re receiving the call with DID +18005551212 and then you play back the ss-noservice.ulaw recording. i’d say it’s not your ISP

3

That was my initial impression too, I am just at a loss as to the reason I cannot for the life of me receive calls on that toll free number. If I receive calls from others on the same provider it should process them the same way.

I will be bald before this all over!

4

This is a 100% you. Twilio sends incoming calls from at least 5 different IPs. Calls to your numbers could come from any of those IPs (it will be in the website FAQs/Docs). Chan_SIP only supports 1 IP per trunk, so you would need a Chan_SIP trunk for each of the IPs Twilio could send calls from.

1 Like
5

This is coming from the “from-sip-external” context, so that’s an incoming (not outgoing) call.

Something in your setup is seriously horked. Time to start at the beginning.

Please anonymize and post your trunk config and outbound route config so we can look at them.

Also, how are these calls getting to the PBX? Normally, this would happen is you were connecting from another PBX and they were trying to call out to the wide world. Something definitely doesn’t add up. The clues point to a single, serious mistake. We should be able to spot it pretty quickly.

6

When I dial the toll free number from my cell phone I receive the ss-noservice. So all of this is based on when I pickup my cell phone and dial the toll free. It is an external call coming into the PBX.

7

And what I said earlier is the answer. Twilio delivers calls from multiple IPs you need a CHAN_SIP trunk for them all.

8

There is no reason to switch to chan_sip. Find out what from which IP addresses Twilio can send calls, then list them in the Match field of your pjsip trunk. You may have to restart Asterisk for the change to take effect.

If you still have trouble, use
pjsip set logger on
to see whether you are receiving the call from an IP address not listed in the Match field.

9

My bad. There is no reason, What he said.

10

Immediate resolution to the problem! Thank you so much.

11

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