@echo501 wrote:

I have 2 extensions out of 18 in FreePBX that will not register with the server. When the extensions were created (over a year ago) they worked fine. We use Grandstream VoIP phones and are a small office. Both phones are GPX1625. All phones and FreePBX server are on the same switch.

Work done so far.

Extension 221

1. Reset SIP password
2. Moved GPX1625 to new network drop
3. Updated Firmware on GPX1625
4. Updated FreePBX modules
5. Factory reset GPX1625 and re-configured
6. Swapped GXP1625 with a working GPX1405 (same results - no registration)
7. Completed a WireShark PCAP capture from the GPX1625. Verified that syn requests are sent to the FreePBX server but there are no responses.

This last step has me questioning why this extension will not register while I have other Grandstream phone that register just fine.

Any thoughts on where to go from here?

Thanks

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@greenftechn wrote:

I have one PSTN line and two SIP trunks. Calls coming in via PSTN can, using follow me, go to my cell phone using a SIP trunk. Calls coming in via SIP, can go out via PSTN. Meanwhile, calls coming in via SIP, while they connect successfully using outbound SIP, no audio is heard at either end.

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@sentinelace wrote:

we have multiple outgoing trunks. The first one when down today so we had no outbound calls. Shouldn’t the next trunk in sequence then start working? I had to manually move it up to the top to make it work.

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@mvogel4949 wrote:

If I use a Snom PA1 to connect to the overhead paging system can I include that extension in the page pro for scheduled pages?

Using a Snom PA1 acts as an extension. I know calling that extension will open the overhead speakers but not sure about including it in a page group

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@gmhicks5867 wrote:

My company is looking at replacing Cisco Communications Manager with FreePBX or another Asterisk platform.

One thing that will be required is to dynamically map and change a phone’s location for E911 purposes. Our Headquarters is in Chicago and Illinois requires E911 zones for every floor and/or 20000 sq. feet. We are currently doing this with Cisco Emergency Responder and doing ERL mappings based on Switch Memberships. Each switch port is assigned to one of our ERLs and the switches communicate with CER via SNMP. when a phone connects to a switch and the Cisco Discovery Protocol (CDP) updates the information the switch sends that information to CER which update’s its database. If that phone moves, or they log in to a different phone then we do not have to manually update the information.

Is this functionality currently available with FreePBX? I am not seeing anything in the documents currently to see that it is.

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@dragonparoxysm wrote:

Now that Sangoma has purchased Digium, what is the plan for Digium phones?
Will they keep making phones under the Digium brand or slowly phase them out?

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@matthewljensen wrote:

My outbound calls are recorded, but they are very slow, so the sound is pitched down. When I download them an increase the speed in VLC, they play ok. This seems to only happen with outbound calls. I’m also using opus, if that might be contributing to the problem. I’m on asterisk 16.0.0.

I found this issue that was closed, but it seems like it is referencing inbound calls, which seem to be OK for me.

What steps should I go through to solve this problem?

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@adelaros wrote:

Hi ,

I am looking for some help regarding a AGI Script that i wrote.

I have created a AGI script that consults a External (FQDN) Web service. A place the agi via Extensions_custom.conf.

exten => 20003,1,Answer
same => n,Set(CHANNEL(language)=es)
same => n,AGI(/var/lib/asterisk/agi-bin/ValeES.php)
same => n,4,Hangup

My problem is that if i call 20003 from a regular extension the Agi work fine , but if do a inbound route from my ITSP and forward the call to 20003 y get the following error from the AGI (debug).

<SIP/TrunkO-00007992>AGI Rx << {“Message”:"No HTTP resource was found that matches the request URI…

Any help is greatly appreciated ,

Thanks,

Alex

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@Matthew99 wrote:

Hello

I have queues set up to fail over to voicemail if call not answered.

This morning I had reported of calls ringing out with noone answering and call not going to voicemail.

After investigating it turns out noone was logged into or available on their phone.

I can kind of see the logic here: if noone logged in call cant enter queue and receive failover instruction but this leaves me with a bit of a problem.

How can I capture calls if noone is logged in?

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@Matthew99 wrote:

Hello all

I seem to be going round in circles here and driving myself slightly mad trying to achieve correct stats which I figured would be fairly simple.

The simple call flow is over with 1 support 2 admin 3 accounts etc if noone answers go to catch all group everyone (basically overflow group of extensions)

I started with ring groups which worked really well in principal using ringall strategy then to second ring group if call not answered.

Then I hit issues trying to produce any useful reports from cdr as calls show it called every extension therefore reporting shows duplicate entries and inaccurate number for amount of calls received. (Happy to provide more info on this If anyone can solve this if anyone has any ideas)

I then started to explore the use of Queues and moved to using Asternic queue reporting. I quickly found issues here with the catch all overflow group as it would only report on the final queue it hit not where it came from. I had to use a little imagination here and created a missed call queue for every equivalent main queue with fsilover to here then voicemail. so now I can track in the report which queue it came from.

This although a little convoluted worked s treat until yesterday I had reports of calls ringing out. This turned out to be because noone was logged into their phone. I think the answer to this is setting strict rule on no queue join which forces to voicemail if users unavailable. The issue here is it then bypasses any queue and therefore doesn’t show on any report.

Someone please help me stop the madness and bring some sanity

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@faisalkhan wrote:

Hi all,

I want to hit a url on incoming calls passing two variables (To and From Number) on call answered.

e.g: http://192.168.0.1/test/asterisk/index.php?from=1237&to=xxx

how can I achieve this. any API required.

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@rnbruce wrote:

I use Queue’s and Queue’s Pro. It seems that the some of the reporting may not be accurate. e.g. when I look at missed call reports I will show an agent missing the same phone number several times in a row. When I look at the cdr it shows no answer for that caller ID several times and then finally shows where the call was answered and points to the same recording on all of the no answers. We are trying to determine what are real missed calls and need the report to be accurate for personnel reasons. Here’s a snip of the cdr:

image.png876×422 119 KB

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@StreetGuru wrote:

Hi,

I have configured an extension used by most incoming trunks with follow me enabled with ringallv2-prim to ring the registered SIP phone and also call my mobile phone;

on the extension follow me list I have:

1001 (the extension)
+1 234 567 890# (my mobile number)

I then have an outbound route to “catch all” to my mobile configured with the dial pattern:

( ) | [+1 234 567 890 / . ]

using another trunk to call.

The set-up works fine except for the fact that I only see the Caller ID configured with the trunk that is being used on the “catch-all” route. How can I configure it so I can see the number of the person calling me?

Thank you in advance for your help.

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@KELSIT wrote:

We have an appliance 100 we bought from the sangoma portal along with 23 s505 phones.

We had test and turn up for our AT&T IP Flex internet and VoIP today. I was told to configure the second interface (eth1) on our appliance with an ip address of 1.1.2.21/24 (I know, that’s not how you’re supposed to use that range, but tell that to AT&T) and gateway of 1.1.2.1. This immediately caused a problem where the system choose that as its default route, which means my phones can’t talk to it and I can’t access the web interface.

After some hurried searching and trial and error, I managed to get the system back to using the other interface (eth0; its address is 10.0.40.9).

I first tried to add 0.0.0.0/0 via 10.0.40.9 dev eth0 to /etc/sysconfig/network-scripts/route-eth0 and re start the network service, but that didn’t work. I also added that same line to /etc/sysconfig/network-scripts/route-eth1, but that didn’t work either. I finally did route add -net 0.0.0.0 netmask 0.0.0.0 gw 10.0.40.254, which worked.
See the resulting route table here:

[root@kfpbx ~]# route -n
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
0.0.0.0         10.0.40.254     0.0.0.0         UG    0      0        0 eth0
0.0.0.0         1.1.2.1         0.0.0.0         UG    0      0        0 eth1
1.1.2.0         0.0.0.0         255.255.255.0   U     0      0        0 eth1
10.0.40.0       0.0.0.0         255.255.255.0   U     0      0        0 eth0
169.254.0.0     0.0.0.0         255.255.0.0     U     1002   0        0 eth0
169.254.0.0     0.0.0.0         255.255.0.0     U     1003   0        0 eth1

That second line concerns me, but the system works; my phones can connect to it and I can access the web interface. I went on to configure the sip trunk to AT&T, but this is where I’m having problems.

The AT&T engineer’s information was basically "send your calls to 1.1.2.1" and not much else. No security, no user account, etc. I fumbled through the pjsip trunk configuration and I’ve gotten to the following situation: I can receive calls via either of the two test phone numbers, but when trying to make a call, I get “all circuits are busy now…”

pjsip settings right now:

image.png1904×3904 323 KB

My outbound routes are the routes for the SIPStation service (I used a trial to test the initial setup of this system), but the trunks set to use the ATT trunk.

Digging into the log files, I think the following lines hint at the problem:

[2019-03-13 20:18:57] ERROR[25471] res_pjsip.c: Endpoint 'att': Could not create dialog to invalid URI 'att'. Is endpoint registered and reachable?
[2019-03-13 20:18:57] ERROR[25471] chan_pjsip.c: Failed to create outgoing session to endpoint 'att'
[2019-03-13 20:18:57] WARNING[20731][C-0000003b] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)

I’ve tried setting the client uri in pjsip settings -> advanced to sip:1.1.2.1:5060, but it still gives me the same log message. I’ve also tried changing the name of the trunk to sip:1.1.2.1:5060, but it just sits quietly until it times out saying that the phone didn’t pick up.

Any ideas?

The full log those 3 lines came from:
(REDACTED is one of the test phone numbers and REDACTED2 is my cell phone that I’m trying to call.

[2019-03-13 20:31:37] VERBOSE[25471] pbx_variables.c: Setting global variable 'SIPDOMAIN' to '10.0.40.9'
[2019-03-13 20:31:37] VERBOSE[25471] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-03-13 20:31:37] VERBOSE[25471] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2019-03-13 20:31:37] VERBOSE[25471] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [REDACTED@from-internal:1] Macro("PJSIP/117-00000054", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/117-00000054", "TOUCH_MONITOR=1552509097.84") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/117-00000054", "AMPUSER=117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/117-00000054", "0?report") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/117-00000054", "1?Set(REALCALLERIDNUM=117)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/117-00000054", "AMPUSER=117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/117-00000054", "0?limit") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/117-00000054", "AMPUSERCIDNAME=Kolton Benoit") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:8] ExecIf("PJSIP/117-00000054", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:9] GotoIf("PJSIP/117-00000054", "0?report") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:10] Set("PJSIP/117-00000054", "AMPUSERCID=117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:11] Set("PJSIP/117-00000054", "__DIAL_OPTIONS=HhTtr") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:12] Set("PJSIP/117-00000054", "CALLERID(all)="Kolton Benoit" <117>") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:13] GotoIf("PJSIP/117-00000054", "0?limit") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:14] ExecIf("PJSIP/117-00000054", "1?Set(GROUP(concurrency_limit)=117)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:15] ExecIf("PJSIP/117-00000054", "0?Set(CHANNEL(language)=)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:16] NoOp("PJSIP/117-00000054", "Macro Depth is 1") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:17] GotoIf("PJSIP/117-00000054", "1?report2:macroerror") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:18] GotoIf("PJSIP/117-00000054", "1?continue") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:37] Set("PJSIP/117-00000054", "CALLERID(number)=117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:38] Set("PJSIP/117-00000054", "CALLERID(name)=Kolton Benoit") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:39] GotoIf("PJSIP/117-00000054", "0?cnum") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:40] Set("PJSIP/117-00000054", "CDR(cnam)=Kolton Benoit") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:41] Set("PJSIP/117-00000054", "CDR(cnum)=117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-user-callerid:42] Set("PJSIP/117-00000054", "CHANNEL(language)=en") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [REDACTED@from-internal:2] Gosub("PJSIP/117-00000054", "sub-record-check,s,1(out,REDACTED2,dontcare)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/117-00000054", "0?initialized") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:2] Set("PJSIP/117-00000054", "__REC_STATUS=INITIALIZED") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:3] Set("PJSIP/117-00000054", "NOW=1552509097") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:4] Set("PJSIP/117-00000054", "__DAY=13") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:5] Set("PJSIP/117-00000054", "__MONTH=03") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:6] Set("PJSIP/117-00000054", "__YEAR=2019") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:7] Set("PJSIP/117-00000054", "__TIMESTR=20190313-203137") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:8] Set("PJSIP/117-00000054", "__FROMEXTEN=117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:9] Set("PJSIP/117-00000054", "__MON_FMT=wav") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/117-00000054", "Recordings initialized") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/117-00000054", "0?Set(ARG3=dontcare)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/117-00000054", "REC_POLICY_MODE_SAVE=") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/117-00000054", "0?Set(REC_STATUS=NO)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/117-00000054", "3?checkaction") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (sub-record-check,s,17)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/117-00000054", "1?sub-record-check,out,1") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (sub-record-check,out,1)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [out@sub-record-check:1] NoOp("PJSIP/117-00000054", "Outbound Recording Check from 117 to REDACTED2") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [out@sub-record-check:2] Set("PJSIP/117-00000054", "RECMODE=dontcare") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [out@sub-record-check:3] ExecIf("PJSIP/117-00000054", "1?Goto(routewins)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (sub-record-check,out,7)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [out@sub-record-check:7] Gosub("PJSIP/117-00000054", "recordcheck,1(dontcare,out,REDACTED2)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/117-00000054", "Starting recording check against dontcare") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/117-00000054", "dontcare") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [recordcheck@sub-record-check:3] Return("PJSIP/117-00000054", "") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [out@sub-record-check:8] Return("PJSIP/117-00000054", "") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [REDACTED2@from-internal:3] ExecIf("PJSIP/117-00000054", "0 ?Set(CDR(accountcode)=)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [REDACTED2@from-internal:4] Set("PJSIP/117-00000054", "MOHCLASS=default") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [REDACTED2@from-internal:5] Set("PJSIP/117-00000054", "_NODEST=") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [REDACTED2@from-internal:6] Macro("PJSIP/117-00000054", "dialout-trunk,1,REDACTED2,,off") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:1] Set("PJSIP/117-00000054", "DIAL_TRUNK=1") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf("PJSIP/117-00000054", "0?Set(DIAL_OPTIONS=Hhtr)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf("PJSIP/117-00000054", "0?sub-pincheck,s,1()") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:4] ExecIf("PJSIP/117-00000054", "0?Set(CALLERID(num)=117)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:5] GotoIf("PJSIP/117-00000054", "0?disabletrunk,1") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:6] Set("PJSIP/117-00000054", "DIAL_NUMBER=REDACTED2") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:7] Set("PJSIP/117-00000054", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:8] Set("PJSIP/117-00000054", "OUTBOUND_GROUP=OUT_1") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:9] Set("PJSIP/117-00000054", "DIAL_TRUNK_OPTIONS=T") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf("PJSIP/117-00000054", "0?nomax") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf("PJSIP/117-00000054", "0?chanfull") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:12] GotoIf("PJSIP/117-00000054", "0?skipoutcid") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:13] Macro("PJSIP/117-00000054", "outbound-callerid,1") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:1] NoOp("PJSIP/117-00000054", "117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:2] NoOp("PJSIP/117-00000054", "") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:3] NoOp("PJSIP/117-00000054", "off") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf("PJSIP/117-00000054", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf("PJSIP/117-00000054", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:6] ExecIf("PJSIP/117-00000054", "0?Set(REALCALLERIDNUM=117)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:7] ExecIf("PJSIP/117-00000054", "0?Set(AMPUSER=117)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:8] GotoIf("PJSIP/117-00000054", "1?normcid") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-outbound-callerid,s,12)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:12] Set("PJSIP/117-00000054", "USEROUTCID=") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:13] Set("PJSIP/117-00000054", "EMERGENCYCID=") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:14] Set("PJSIP/117-00000054", "TRUNKOUTCID=REDACTED") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:15] GotoIf("PJSIP/117-00000054", "1?trunkcid") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-outbound-callerid,s,20)
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:20] ExecIf("PJSIP/117-00000054", "1?Set(CALLERID(all)=REDACTED)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf("PJSIP/117-00000054", "0?Set(CALLERID(all)=)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf("PJSIP/117-00000054", "0?Set(CALLERID(all)=)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf("PJSIP/117-00000054", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:24] ExecIf("PJSIP/117-00000054", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:25] Set("PJSIP/117-00000054", "CDR(outbound_cnum)=REDACTED") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outbound-callerid:26] Set("PJSIP/117-00000054", "CDR(outbound_cnam)=") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:14] GosubIf("PJSIP/117-00000054", "0?sub-flp-1,s,1()") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:15] Set("PJSIP/117-00000054", "OUTNUM=REDACTED2") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:16] Set("PJSIP/117-00000054", "custom=PJSIP") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf("PJSIP/117-00000054", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf("PJSIP/117-00000054", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:19] Macro("PJSIP/117-00000054", "dialout-trunk-predial-hook,") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/117-00000054", "") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:20] GotoIf("PJSIP/117-00000054", "0?skipcrm") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:21] Set("PJSIP/117-00000054", "__CRM_DIRECTION=OUTBOUND") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:22] Set("PJSIP/117-00000054", "__CRM_DESTINATION=REDACTED2") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:23] Set("PJSIP/117-00000054", "__CRM_SOURCE=117") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:24] AGI("PJSIP/117-00000054", "sangomacrm.agi") in new stack
[2019-03-13 20:31:37] VERBOSE[30088][C-0000003d] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] res_agi.c: <PJSIP/117-00000054>AGI Script sangomacrm.agi completed, returning 0
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:25] Set("PJSIP/117-00000054", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp("PJSIP/117-00000054", "CRM Finished") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf("PJSIP/117-00000054", "0?bypass,1") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf("PJSIP/117-00000054", "1?Set(CONNECTEDLINE(num,i)=REDACTED2)") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf("PJSIP/117-00000054", "1?Set(CONNECTEDLINE(name,i)=CID:REDACTED)") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf("PJSIP/117-00000054", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)REDACTED)") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf("PJSIP/117-00000054", "0?customtrunk") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:32] Dial("PJSIP/117-00000054", "PJSIP/REDACTED2@att,300,Tb(func-apply-sipheaders^s^1,(1))") in new stack
[2019-03-13 20:31:38] ERROR[27195] res_pjsip.c: Endpoint 'att': Could not create dialog to invalid URI 'att'. Is endpoint registered and reachable?
[2019-03-13 20:31:38] ERROR[27195] chan_pjsip.c: Failed to create outgoing session to endpoint 'att'
[2019-03-13 20:31:38] WARNING[30088][C-0000003d] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:33] NoOp("PJSIP/117-00000054", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-dialout-trunk:34] GotoIf("PJSIP/117-00000054", "0?continue,1:s-CHANUNAVAIL,1") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("PJSIP/117-00000054", "RC=3") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("PJSIP/117-00000054", "3,1") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-dialout-trunk,3,1)
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [3@macro-dialout-trunk:1] Goto("PJSIP/117-00000054", "continue,1") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-dialout-trunk,continue,1)
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp("PJSIP/117-00000054", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 3 - failing through to other trunks") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [continue@macro-dialout-trunk:2] ExecIf("PJSIP/117-00000054", "1?Set(CALLERID(number)=117)") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [REDACTED2@from-internal:7] Macro("PJSIP/117-00000054", "outisbusy,") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outisbusy:1] Progress("PJSIP/117-00000054", "") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outisbusy:2] GotoIf("PJSIP/117-00000054", "0?emergency,1") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outisbusy:3] GotoIf("PJSIP/117-00000054", "0?intracompany,1") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-outisbusy:4] Playback("PJSIP/117-00000054", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
[2019-03-13 20:31:38] VERBOSE[30088][C-0000003d] file.c: <PJSIP/117-00000054> Playing 'all-circuits-busy-now.g722' (language 'en')
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] file.c: <PJSIP/117-00000054> Playing 'please-try-call-later.ulaw' (language 'en')
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] pbx.c: Executing [h@from-internal:1] Macro("PJSIP/117-00000054", "hangupcall") in new stack
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/117-00000054", "1?theend") in new stack
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("PJSIP/117-00000054", "0?Set(CDR(recordingfile)=)") in new stack
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-hangupcall:4] NoOp("PJSIP/117-00000054", " monior file= ") in new stack
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-hangupcall:5] AGI("PJSIP/117-00000054", "attendedtransfer-rec-restart.php,,") in new stack
[2019-03-13 20:31:39] VERBOSE[30088][C-0000003d] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] res_agi.c: <PJSIP/117-00000054>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@macro-hangupcall:6] Hangup("PJSIP/117-00000054", "") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'PJSIP/117-00000054' in macro 'hangupcall'
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/117-00000054'
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] app_stack.c: PJSIP/117-00000054 Internal Gosub(crm-hangup,s,1) start
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:1] NoOp("PJSIP/117-00000054", "Sending Hangup to CRM") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:2] NoOp("PJSIP/117-00000054", "HANGUP CAUSE: 3") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:3] ExecIf("PJSIP/117-00000054", "0?Set(__CRM_VOICEMAIL=)") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:4] NoOp("PJSIP/117-00000054", "MASTER CHANNEL: 1552509097.84 = 1552509097.84") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:5] GotoIf("PJSIP/117-00000054", "0?return") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:6] Set("PJSIP/117-00000054", "__CRM_HANGUP=1") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:7] AGI("PJSIP/117-00000054", "sangomacrm.agi") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] res_agi.c: <PJSIP/117-00000054>AGI Script sangomacrm.agi completed, returning 0
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] pbx.c: Executing [s@crm-hangup:8] Return("PJSIP/117-00000054", "") in new stack
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/117-00000054'
[2019-03-13 20:31:40] VERBOSE[30088][C-0000003d] app_stack.c: PJSIP/117-00000054 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

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@arjones5 wrote:

We are adding Yealink wireless W60B/W56H DECT phones to our deployment and I’m coming across an issue that I’ve never encountered. Given that I’ve read through almost all the related forums and there are many of you that rave about these phones…I need help. I cannot get the accounts to register. I set the SIP server as the freePBX host on port 5160 and all it throws back is “Register Fail”.

For anyone out there that has set these up before, do you have any tips or hints for setting these up?

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@Eko_VoIP wrote:

Hello Community members,

I have installed FreePBX and Digium A8B analog Card- 8 Port FXO PCI Express on a Dell PC and it’s workign fine, but i am wondering if it is possible the same card type to install and use if i buy Sangoma PBXact 100 ?, does anybody did it before or just need to buy sangoma card with PBXact products?
I read on this post about supported cards from FreePBX but I am not sure about PBXact?
https://wiki.freepbx.org/display/FPG/Supported+Cards

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@gtrovato wrote:

Hi All,

I’ve a strange behavior on my AsteriskNow (FreePBX 13.0.195.18 - Asterisk 13.12.1).
I’ve a setup with 2 SIP trunks (Grandstream and Lynksys ATA) and IVR.
All my setup is standard SIP (no PJSIP).
Sometimes, during a call, the IVR starts in background; it’s audible from both peers.

What could be the cause?

Thank you!

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@phicinque wrote:

I have FreePBX 14.0.5.25 running on Ubuntu 18.04.

After an upgrade of Pm2, the command fwconsole has stopped working.

When I try “fwconsole ma …” I have always the message:
“Bug: Explicitly know about FreePBX\Console\Command, asked for FreePBX\Console\Command\Pm2, but /var/www/html/admin/libraries/Console/Pm2.php (or Pm2.class.php) doesn’t exist”

Asterisk is running.
The GUI, after login, goes in error:
"Whoops \ Exception \ ErrorException (E_WARNING)
simplexml_load_file(): /var/www/html/admin/modules/sipstation/module.xml:1: parser error : Document is empty "

What can I do?
Any help will be appreciate

Thanks,
phicinque

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@wzkds wrote:

I saw a couple of old posts with this question, but no answer to the question. I’d like to be able to see which users are logged into which devices that are in adhoc mode.

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@anscomputer wrote:

Hello,

i need to configure that :
a customer call the number.
my phone ring during 15sec if nobody unhook there is an announcement.

Is it possible to unhook during the announcement ?
(I have configured a queue with an annoucement when you enter the queue)

Best regards,
Luca

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