So, if you make an outbound call, the system works as long as the call comes in within 30 seconds? Audio all works, call terminates in the PBX and is connected to a local phone or service?

I’m still thinking it’s a problem with your router not passing the traffic from your external address to the PBX because, well, that’s almost always what causes this. Do you have all of the SIP addresses set correctly?

If you look back through the forum, you’ll find lots of cases where everything works for about 30 seconds, and the problems are almost always a problem at the router.

It is an interesting conversation. The way I see things is Asterisk is a good product I used for the first time in 2009 but was only using it as a conference bridge attached to my company’s NEC Proprietary digital, Analog and VoIP Ports added. Had I have been aware of Asterisk before getting the NEC I would have Never purchased the NEC as it was a disaster. Many features did not work as promised. My company was not integrating Phones of any kind at that time. But due to the issues with NEC It was part of my long-term goal to get into telephony. So here we are. New to Integrating telephony or these days VoIP is the most common term.
Since we were a specialized network company we did do some work on Texas A&M’s install of Asterisk in I think 07. Over 5k extensions with many of them analog phones.
We started studying the various solutions about 1 year ago. At that time 3CX was Windows Only. I certainly will NOT use a Windows-based PBX unless a customer insists on it and so far this has not happened. All systems will have some strong points and some weak areas. That is just a fact. But 3CX cost us a very large sale and a few smaller installs. We were given very wrong info by the Staff. I will not go into detail but I have zero tolerance for a few things they did. Maybe FreePBX is not perfect Nothing is. BUT I found good folks and had my eye on FreePBX and a couple other solutions. But FreePBX has the features we feel our customers will want. And Sangoma has some good looking solutions that were clearly well thought out to have what is needed and NOT overloaded with a bunch of stuff we do NOT need. Plus it looks Professional. This can be important for some clients. I do not wish badly on 3CX although I certainly could justify doing so.
On another note I am pleased there is an ARM based version of FreePBX as 1. I think ARM will at some point have a bigger role in PBXes at some future Point. 2. RISC chips are in my experience superior to CISC with this type of software. However, I personally will NOT use an experiment board such as the Raspberry Pi as a PBX server even as cool as they are. The CPU is fine but these are NOT commercial grade boards. I figure boards with a more reliable PCB Other than cell phones and Google chrome books will be available at some time and Sangoma will integrate one of them. They are great to play with though for now.
You folks are an interesting bunch and we look forward to working with you folks.
Jim

First thread. Excuse any breech of etiquette or omitting of accepted nomenclature.

FreePBX 14.0.3.6
PBX Firmware:12.7.5-1807-1.sng7
PBX Service Pack:1.0.0.0

Having intermittent call failure on 2 DID’s inbound. Many times it works fine.

PBX is cloud hosted

I believe that the Inbound Edge Strategy or “PoP” SIP servers for load balancing are not consistently resolving to my PBX. But I am not sure…

I have white listed all of the proposed PoP subnets from FlowRoute in the sysadmin module:

147.75.65.192/28
147.75.60.160/28
34.226.36.32/28
34.210.91.112/28

I have also trusted them within the firewall in FreePBX module in the appropriate zone. (Note: this is the only firewall.)

I feel like the problem could exist withing the srv lookup that they are using with their Pop servers. They don’t offer a great deal of elaboration on trunk configuration:

https://support.flowroute.com/SIP_Trunking_and_Voice/PBX__Configuration_Guides/Asterisk/Configure_Asterisk_13

Checking in my Fail2Ban log, I see that:

[2018-08-10 10:52:45] WARNING[20377][C-00000166] Ext. s: "Rejecting unknown SIP connection from 34.210.91.114 (This is one of the whitelisted ip addresses mentioned earlier)

That address was specifically allowed through the firewall in the trusted traffic under the 34.210.91.112/28 addition.

Also, here is the output from a call where I received "the number you have dialed is not in service."

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7efc50046d90 – Strict RTP learning after remote address set to: 74.1 20.93.196:21466
– Executing [12082058780@from-sip-external:1] NoOp(“SIP/fl.gg-000001d9”, “R eceived incoming SIP connection from unknown peer to 12082058780”) in new stack
– Executing [12082058780@from-sip-external:2] Set(“SIP/fl.gg-000001d9”, “DI D=12082058780”) in new stack
– Executing [12082058780@from-sip-external:3] Goto(“SIP/fl.gg-000001d9”, “s ,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/fl.gg-000001d9”, “1?setlang uage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“SIP/fl.gg-000001d9”, “CHANNEL(lang uage)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“SIP/fl.gg-000001d9”, “1?noanony mous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“SIP/fl.gg-000001d9”, “TIMEOUT(abso lute)=15”) in new stack
– Channel will hangup at 2018-08-10 11:34:47.685 MDT.
– Executing [s@from-sip-external:6] Log(“SIP/fl.gg-000001d9”, "WARNING,“Rej ecting unknown SIP connection from 34.210.91.114"”) in new stack
[2018-08-10 11:34:32] WARNING[25179][C-00000168]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 34.210.91.114"
– Executing [s@from-sip-external:7] Answer(“SIP/fl.gg-000001d9”, “”) in new stack
> 0x7efc50046d90 – Strict RTP switching to RTP target address 74.120.93. 196:21466 as source
– Executing [s@from-sip-external:8] Wait(“SIP/fl.gg-000001d9”, “2”) in new stack
– Executing [s@from-sip-external:9] Playback(“SIP/fl.gg-000001d9”, “ss-nose rvice”) in new stack
– <SIP/fl.gg-000001d9> Playing ‘ss-noservice.ulaw’ (language ‘en’)
> 0x7efc50046d90 – Strict RTP learning complete - Locking on source addr ess 74.120.93.196:21466
– Executing [h@from-sip-external:1] Hangup(“SIP/fl.gg-000001d9”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/fl.gg-000 001d9’

If I have allowed the traffic through the firewall, and it is white listed, why would it block the IP???

Was hoping someone would offer helpful suggestions. Hint’s on configuring Flowroute trunks with srv lookup?

Thank you

I have port 5060 and 10000-20000 forwarding from twilio’s list of IP addresses to our PBX internal IP. I have the pbx firewall off for testing purposes.

There are 0 denials in the firewall log.

We are running FreePBX 13.0.195.4 and we are running this in a test environment. We use Twilio for our SIP trunking. Normally we would use xxx1.pstn.us1.twilio.com for our termination URI defined in the outgoing peer details of the trunk.

This is a system we had in production but now we just want to test in our test environment so we reset the termination URI to xxx-lab.pstn.us1.twilio.com. When we try to make a call from the server to an outside number, we get the following error from Twilio.
Authentication failure - source IP Address not in ACL.
When I look at the details, it’s showing that the to: is the number I’m dialing @ xxx.ptsn.us1.twilio.com (the old termination URI) and then the from is CID@10.0.1.36.

1. Why is it showing an internal address?
2. Why is it using the wrong termination URI? I can’t find anywhere that it would still be defined. The trunk outgoing SIP peer details is the only place that I know it’s defined and it’s setup for the lab URI.

When I call out, I get the “all circuits are busy” message.

Any suggestions?

Hello,

I just thought I would put a query out there for what folks might recommend for using as a soft phone for a receptionist/operator? Paid or free is fine. The client has FOP2 in use as well.

Receptionist/operator currently has a desk phone (Aastra 6867i + 2 expansion modules), but due to what appear to be hardware limitations, only the first 50 or so BLF/Xfer entries on the expansion modules actually show proper presence (the rest just show a ‘?’)

I am thinking that since they have FOP2 in use anyway (currently with a touch screen, but I think it would probably be way more useful with a mouse), and the page is access via a dedicated PC at the receptionst’s desk, it might be better to do away with the physical phone and just install a decent soft phone on the dedicated PC and use a head set.

I am not sure if doing that it would still be possible to do transfers, etc. through FOP2? I imagine it would, but just thought I would check.

I am aware of a few different soft phones, but wanted to see what the recommended one might be specifically for a receptionist/operator?

Suggestions are welcome!

I wouldn’t rely on SRV working reliably with chan_sip trunks. You will probably have better luck using Asterisk 15 and PJSIP trunks, or you can create a separate chan_sip trunk for all of the provider’s signalling hosts.

I actually tried deleting the extension, applying the config, the adding the extension back, but the problem persisted. I am almost certain it has to do with how the trunk provider is passing the CID to us.

I will give that a shot. I saw that was what some people were doing.

This is FlowRoute’s note:

IMPORTANT: Asterisk for Flowroute with New PoPs

If you are taking advantage of our new Points of Presence (PoPs), make sure to do the following:

1. Update all instances of sip.flowroute.com to this format: {your_preferred_pop}.sip.flowroute.com where {your_preferred_pop} might be “us-west-wa” for example. The new value will then be us-west-wa.sip.flowroute.com .

2. Add the IP addresses associated with your preferred PoP to your list of [flowroute-trunk] settings below. To find this information quickly, run the following from a command shell window:

dig {your_preferred_pop}.sip.flowroute.com +short

Did an fwconsole ma list, and created scripts to download and install all of my enabled modules. This did nothing to the epm. Not sure what to try next; if I restore my system’s vm from last week before I made the changes, it will appear as new hardware and I’ll have a big headache with Zend resets and commercial licensing.

Ideas I could check?

Which endpoint version do you have installed?

Did you try Zulu? It has a free Trial.

Can you post a failed call log?

Michael,

I dont mean to be rude by any means, however did you even read my post before posting. lol. It says " S705 Phones (which are Sangoma IP Phones) " and “When calling in or dialing on the phone pad itself they are “static-y” when dialing or ringing” (which means when the phone is ringing or the sound the phone makes when dialing on the dial pad without picking up the handset). This is a very standard setup with no modifications to any base files, using all ring defaults and all codec defaults. There is 5 phones in the one deployment and 20 phones in the other. They are ALL doing it.

Otherwise I agree, its normally a cable issue which i have encountered. I know static is from loose audio connections. However this is why i put “Static-y” in quotes because its hard to describe it.

Like if you go into the menu and select different ring tones they sound correct. However when the phone is ACTUALLY ringing its all cut in and out kinda like its a loose connection. They all do it, its very odd. And its not just this deployment its another one with a totally different environment. However I can say this, they all came from the same batch.

I would contact support, I assume there’s some background applications that were sleeping.

Thanks Lorne,
The link isn’t in the settings menu either. I only have “CRM settings”

[2018-08-13 15:13:40] VERBOSE[5511][C-00000082] netsock2.c: Using SIP RTP TOS bits 184
[2018-08-13 15:13:40] VERBOSE[5511][C-00000082] netsock2.c: Using SIP RTP CoS mark 5
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [7025242574@from-internal:1] Macro("SIP/202-00000086", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:1] Set("SIP/202-00000086", "TOUCH_MONITOR=1534198420.139") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:2] Set("SIP/202-00000086", "AMPUSER=202") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("SIP/202-00000086", "0?report") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("SIP/202-00000086", "1?Set(REALCALLERIDNUM=202)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:5] Set("SIP/202-00000086", "AMPUSER=202") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("SIP/202-00000086", "0?limit") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:7] Set("SIP/202-00000086", "AMPUSERCIDNAME=Todd G. Schacherl") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:8] ExecIf("SIP/202-00000086", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:9] GotoIf("SIP/202-00000086", "0?report") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:10] Set("SIP/202-00000086", "AMPUSERCID=202") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:11] Set("SIP/202-00000086", "__DIAL_OPTIONS=Ttr") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:12] Set("SIP/202-00000086", "CALLERID(all)="Todd G. Schacherl" <202>") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:13] GotoIf("SIP/202-00000086", "0?limit") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:14] ExecIf("SIP/202-00000086", "1?Set(GROUP(concurrency_limit)=202)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:15] ExecIf("SIP/202-00000086", "0?Set(CHANNEL(language)=)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:16] NoOp("SIP/202-00000086", "Macro Depth is 1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:17] GotoIf("SIP/202-00000086", "1?report2:macroerror") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:18] GotoIf("SIP/202-00000086", "1?continue") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:37] Set("SIP/202-00000086", "CALLERID(number)=202") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:38] Set("SIP/202-00000086", "CALLERID(name)=Todd G. Schacherl") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:39] GotoIf("SIP/202-00000086", "0?cnum") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:40] Set("SIP/202-00000086", "CDR(cnam)=Todd G. Schacherl") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:41] Set("SIP/202-00000086", "CDR(cnum)=202") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-user-callerid:42] Set("SIP/202-00000086", "CHANNEL(language)=en") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [7025242574@from-internal:2] Gosub("SIP/202-00000086", "sub-record-check,s,1(out,7025242574,dontcare)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:1] GotoIf("SIP/202-00000086", "0?initialized") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:2] Set("SIP/202-00000086", "__REC_STATUS=INITIALIZED") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:3] Set("SIP/202-00000086", "NOW=1534198420") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:4] Set("SIP/202-00000086", "__DAY=13") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:5] Set("SIP/202-00000086", "__MONTH=08") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:6] Set("SIP/202-00000086", "__YEAR=2018") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:7] Set("SIP/202-00000086", "__TIMESTR=20180813-151340") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:8] Set("SIP/202-00000086", "__FROMEXTEN=202") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:9] Set("SIP/202-00000086", "__MON_FMT=wav") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:10] NoOp("SIP/202-00000086", "Recordings initialized") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:11] ExecIf("SIP/202-00000086", "0?Set(ARG3=dontcare)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:12] Set("SIP/202-00000086", "REC_POLICY_MODE_SAVE=") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:13] ExecIf("SIP/202-00000086", "0?Set(REC_STATUS=NO)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:14] GotoIf("SIP/202-00000086", "3?checkaction") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-record-check:17] GotoIf("SIP/202-00000086", "1?sub-record-check,out,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (sub-record-check,out,1)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [out@sub-record-check:1] NoOp("SIP/202-00000086", "Outbound Recording Check from 202 to 7025242574") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [out@sub-record-check:2] Set("SIP/202-00000086", "RECMODE=force") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [out@sub-record-check:3] ExecIf("SIP/202-00000086", "0?Goto(routewins)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [out@sub-record-check:4] ExecIf("SIP/202-00000086", "0?Goto(routewins)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [out@sub-record-check:5] Gosub("SIP/202-00000086", "recordcheck,1(force,out,7025242574)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("SIP/202-00000086", "Starting recording check against force") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("SIP/202-00000086", "force") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (sub-record-check,recordcheck,5)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:5] Set("SIP/202-00000086", "__REC_POLICY_MODE=FORCE") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:6] GotoIf("SIP/202-00000086", "1?startrec") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (sub-record-check,recordcheck,16)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:16] NoOp("SIP/202-00000086", "Starting recording: out, 7025242574") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:17] Set("SIP/202-00000086", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:18] Set("SIP/202-00000086", "__CALLFILENAME=out-7025242574-202-20180813-151340-1534198420.139") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:19] MixMonitor("SIP/202-00000086", "2018/08/13/out-7025242574-202-20180813-151340-1534198420.139.wav,abi(LOCAL_MIXMON_ID),") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:20] Set("SIP/202-00000086", "__MIXMON_ID=0x7f91a120ebf0") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:21] Set("SIP/202-00000086", "__RECORD_ID=SIP/202-00000086") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:22] Set("SIP/202-00000086", "__REC_STATUS=RECORDING") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:23] Set("SIP/202-00000086", "CDR(recordingfile)=out-7025242574-202-20180813-151340-1534198420.139.wav") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [recordcheck@sub-record-check:24] Return("SIP/202-00000086", "") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [out@sub-record-check:6] Return("SIP/202-00000086", "") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [7025242574@from-internal:3] ExecIf("SIP/202-00000086", "0 ?Set(CDR(accountcode)=)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [7025242574@from-internal:4] Set("SIP/202-00000086", "MOHCLASS=default") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [7025242574@from-internal:5] Set("SIP/202-00000086", "_NODEST=") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [7025242574@from-internal:6] Macro("SIP/202-00000086", "dialout-trunk,6,17025242574,,off") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:1] Set("SIP/202-00000086", "DIAL_TRUNK=6") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf("SIP/202-00000086", "0?Set(DIAL_OPTIONS=tr)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf("SIP/202-00000086", "0?sub-pincheck,s,1()") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:4] GotoIf("SIP/202-00000086", "0?disabletrunk,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:5] Set("SIP/202-00000086", "DIAL_NUMBER=17025242574") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:6] Set("SIP/202-00000086", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:7] Set("SIP/202-00000086", "OUTBOUND_GROUP=OUT_6") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:8] Set("SIP/202-00000086", "DIAL_TRUNK_OPTIONS=Tt") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf("SIP/202-00000086", "1?nomax") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-dialout-trunk,s,11)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf("SIP/202-00000086", "0?skipoutcid") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:12] Macro("SIP/202-00000086", "outbound-callerid,6") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:1] ExecIf("SIP/202-00000086", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:2] ExecIf("SIP/202-00000086", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:3] ExecIf("SIP/202-00000086", "0?Set(REALCALLERIDNUM=202)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf("SIP/202-00000086", "0?Set(AMPUSER=202)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:5] GotoIf("SIP/202-00000086", "1?normcid") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-outbound-callerid,s,9)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:9] Set("SIP/202-00000086", "USEROUTCID=") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:10] Set("SIP/202-00000086", "EMERGENCYCID=") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:11] Set("SIP/202-00000086", "TRUNKOUTCID=+17029860371") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:12] GotoIf("SIP/202-00000086", "1?trunkcid") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-outbound-callerid,s,17)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:17] ExecIf("SIP/202-00000086", "1?Set(CALLERID(all)=+17029860371)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:18] ExecIf("SIP/202-00000086", "0?Set(CALLERID(all)=)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:19] ExecIf("SIP/202-00000086", "0?Set(CALLERID(all)=)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:20] ExecIf("SIP/202-00000086", "0?Set(CALLERID(all)=)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf("SIP/202-00000086", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf("SIP/202-00000086", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:23] Set("SIP/202-00000086", "CDR(outbound_cnum)=+17029860371") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outbound-callerid:24] Set("SIP/202-00000086", "CDR(outbound_cnam)=") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:13] GosubIf("SIP/202-00000086", "1?sub-flp-6,s,1()") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@sub-flp-6:1] ExecIf("SIP/202-00000086", "1?Return()") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:14] Set("SIP/202-00000086", "OUTNUM=+17025242574") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:15] Set("SIP/202-00000086", "custom=SIP/TwilioSIP") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf("SIP/202-00000086", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf("SIP/202-00000086", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:18] Macro("SIP/202-00000086", "dialout-trunk-predial-hook,") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/202-00000086", "") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:19] GotoIf("SIP/202-00000086", "0?skipcrm") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:20] Set("SIP/202-00000086", "__CRM_DIRECTION=OUTBOUND") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:21] Set("SIP/202-00000086", "__CRM_DESTINATION=+17025242574") in new stack
[2018-08-13 15:13:40] VERBOSE[28682][C-00000082] app_mixmonitor.c: Begin MixMonitor Recording SIP/202-00000086
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:22] Set("SIP/202-00000086", "__CRM_SOURCE=202") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:23] AGI("SIP/202-00000086", "sangomacrm.agi") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] res_agi.c: <SIP/202-00000086>AGI Script sangomacrm.agi completed, returning 0
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:24] Set("SIP/202-00000086", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:25] NoOp("SIP/202-00000086", "CRM Finished") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:26] GotoIf("SIP/202-00000086", "0?bypass,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:27] ExecIf("SIP/202-00000086", "1?Set(CONNECTEDLINE(num,i)=17025242574)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf("SIP/202-00000086", "1?Set(CONNECTEDLINE(name,i)=CID:+17029860371)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf("SIP/202-00000086", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)+17029860371)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:30] GotoIf("SIP/202-00000086", "0?customtrunk") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:31] Dial("SIP/202-00000086", "SIP/TwilioSIP/+17025242574,300,Tt") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] netsock2.c: Using SIP RTP TOS bits 184
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] netsock2.c: Using SIP RTP CoS mark 5
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] app_dial.c: Called SIP/TwilioSIP/+17025242574
[2018-08-13 15:13:40] WARNING[5511][C-00000082] chan_sip.c: Received response: "Forbidden" from '<sip:+17029860371@206.198.135.252>;tag=as3cec8aeb'
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:32] NoOp("SIP/202-00000086", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-dialout-trunk:33] GotoIf("SIP/202-00000086", "0?continue,1:s-CHANUNAVAIL,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/202-00000086", "RC=21") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/202-00000086", "21,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-dialout-trunk,21,1)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [21@macro-dialout-trunk:1] Goto("SIP/202-00000086", "continue,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-dialout-trunk,continue,1)
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp("SIP/202-00000086", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/202-00000086", "1?Set(CALLERID(number)=202)") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [7025242574@from-internal:7] Macro("SIP/202-00000086", "outisbusy,") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outisbusy:1] Progress("SIP/202-00000086", "") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outisbusy:2] GotoIf("SIP/202-00000086", "0?emergency,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outisbusy:3] GotoIf("SIP/202-00000086", "0?intracompany,1") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-outisbusy:4] Playback("SIP/202-00000086", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
[2018-08-13 15:13:40] VERBOSE[28675][C-00000082] file.c: <SIP/202-00000086> Playing 'all-circuits-busy-now.ulaw' (language 'en')
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] file.c: <SIP/202-00000086> Playing 'please-try-call-later.ulaw' (language 'en')
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [h@from-internal:1] Macro("SIP/202-00000086", "hangupcall") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/202-00000086", "1?theend") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/202-00000086", "0?Set(CDR(recordingfile)=)") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/202-00000086", "") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/202-00000086' in macro 'hangupcall'
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/202-00000086'
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] app_stack.c: SIP/202-00000086 Internal Gosub(crm-hangup,s,1) start
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:1] NoOp("SIP/202-00000086", "Sending Hangup to CRM") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:2] NoOp("SIP/202-00000086", "HANGUP CAUSE: 21") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:3] ExecIf("SIP/202-00000086", "0?Set(__CRM_VOICEMAIL=)") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:4] NoOp("SIP/202-00000086", "MASTER CHANNEL: 1534198420.139 = 1534198420.139") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:5] GotoIf("SIP/202-00000086", "0?return") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:6] Set("SIP/202-00000086", "__CRM_HANGUP=1") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:7] AGI("SIP/202-00000086", "sangomacrm.agi") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] res_agi.c: <SIP/202-00000086>AGI Script sangomacrm.agi completed, returning 0
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] pbx.c: Executing [s@crm-hangup:8] Return("SIP/202-00000086", "") in new stack
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/202-00000086'
[2018-08-13 15:13:42] VERBOSE[28675][C-00000082] app_stack.c: SIP/202-00000086 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2018-08-13 15:13:42] VERBOSE[28682][C-00000082] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2018-08-13 15:13:42] VERBOSE[28682][C-00000082] app_mixmonitor.c: End MixMonitor Recording SIP/202-00000086

[2018-08-13 15:13:40] WARNING[5511][C-00000082] chan_sip.c: Received response: “Forbidden” from ‘&lt;sip:+17029860371@206.198.135.252&gt;;tag=as3cec8aeb’

apparently you can’t send calls through 206.198.135.252

RIght, I get that. It’s trying to send to cfusa-cc.pstn.us1.twilio.com when it should be sending to cclab.ptsn.us1.twilio.com. What I don’t get is why it’s still referencing cfusa-cc when it says cclab in the outbound peer settings for the Twilio trunk.

IOW, cfusa-cc has whitelisted our .250 addr and cclab has .252 whitelisted. Why is it trying to hit cfusa-cc instead of cclab?