@robinsonjas wrote:

Deployment w/ latest ISO & all patches…Setup Paging Pro
I have 6 endpoints that the paging group will ‘ring’ for the bell schedule but the endpoints are stuck ‘In Use’ until I force hangup. Anyone else have issues like this?
If I just call an endpoint it doesn’t hang when I just intercom to it.

Posts: 2

Participants: 2

Read full topic

@aspieboy77 wrote:

Hello i seam to be having some issues with my admin ui and trying to load the interface it will error out and chrome will say the page took too long to load and i have to restart the vm to get it to run again and then it only works for 2 minutes or so if someone could please help me with this that would be grate thanks

Posts: 2

Participants: 1

Read full topic

@wassy83 wrote:

Hi to all,
just searching a little help to understand something that happens recently in my production environment, sorry for the long post, just wanna add more details as possible.

First of all I have a freepbx kvm inside a big HA proxmox cluster, in the same cluster is present another PFsense KVM that I use for networking stuffs and as gateway for freepbx and all the other VMs and workstations.
Behind this cluster there is a super fast VDSL 300/30Mb that is combined with a fritzbox router(this is the provider’s router and I cannot change it or I will loose internet connectivity) so freepbx is behind double NAT.

• Frepbx and pfsense are latest.
• I have 4 pjsip trunks to the Italian provider EHIWEB.
• CHAN_SIP is disabled so only CHAN_PJSIP DRIVER is available.
• my DNS are 127.0.0.1,8.8.8.8, 8.8.4.4
• I have around 100 pjsip extensions that are all local or through an IPSEC VPN.
• I have a public static IP.
• Since all extensions are local I don’t have forwarded any ports to freepbx.

Everything works perfect, I didn’t have any issue for 2 years(except some problems related to the voip provider)
in a massive production stage(30/35 external calls together plus all local traffic).

But in the last mounth something is happening and I didn’t understand why, aleatory I cannot ping anymore my voip provider IP voip.vivavox.it so trunks goes offline and to fix this I have to reboot. In that moment I can ping any other IP and from local LAN I can ping voip.vivavox.it with no problems, so is related only to freepbx. Anyway even if is a little anooying is not so bad to reboot freepbx sometime but this friday things went really wrong:

the node that is hosting FREEPBX and PFSENSE shutted down due an hardware failure but my proxmox cluster did things well and migrated the freepbx KVM and pfsense KVM from the faulty node to an online node, but obviously this wasn’t a live migration so the freepbx VM and Pfsense VM received an hard reset and then they came back online again. HERE COMES THE PROBLEMS again I was able to ping everything but not voip.vivavox.it and obviously all my trunks were unregistered receiving error “no response from voip.vivavox.it”, so I rebooted again in a safe way pfsense and freepbx and ping was working, the trunks turn back online but I have no bidirectional voice, I rebooted, disabled firewall, disabled fail2ban but with no success so production went down. I tried to register to my trunks with a softphone from the same pfsense connectivity and it was working fine. No way to fix this, after let’s say 1 hour I rebooted again freepbx and then everything worked fine. I really can’t understand why this happens, in particular why at a certain point I cannot ping anymore voip.vivavox,it. I called the voip provider(just in case) and they told to me that there wasn’t any particular issue in that moment.
I think that for some reason my freepbx locks the connection with this particular DNS,
if this help I noticed lately that when I send a netcat to this DOMAIN from my freepbx I’m receiving this

root@freepbx:~# netcat -u -vv voip.vivavox.it 5060
DNS fwd/rev mismatch: voip.vivavox.it != voip.eutelia.it
voip.vivavox.it [83.211.227.21] 5060 (sip) open

or with ping

[root@freepbx ~]# ping voip.vivavox.it
PING voip.vivavox.it (83.211.227.21) 56(84) bytes of data.
64 bytes from voip.eutelia.it (83.211.227.21): icmp_seq=1 ttl=55 time=22.3 ms
64 bytes from voip.eutelia.it (83.211.227.21): icmp_seq=2 ttl=55 time=21.1 ms
64 bytes from voip.eutelia.it (83.211.227.21): icmp_seq=3 ttl=55 time=21.2 ms
64 bytes from voip.eutelia.it (83.211.227.21): icmp_seq=4 ttl=55 time=21.1 ms
^X^C
— voip.vivavox.it ping statistics —
4 packets transmitted, 4 received, 0% packet loss, time 3280ms
rtt min/avg/max/mdev = 21.153/21.459/22.319/0.496 ms

I’M PRETTY SURE THAT THIS MISMATCH WASN’T HAPPENING ONE MOUNTH AGO

Any suggestion on what I can try if this happens again?
many thanks

Posts: 2

Participants: 2

Read full topic

@betabimmer wrote:

I am having a problem to configure something with Dids so let me first explain my problem:

I have a TRUNK with sip username/password so it gives me only digest auth not ip authentication and this trunk have 15 dids so if I login into it from my computer directly with phonerlite or eyebeam any number from this 15 dids called it will ring and I can answer.

Now I want to use this dids from FreePBX register this trunk in FreePBX and there are some agents which everyone have a speicific extension and point one did from those 15 to each of those agents/extensions

Is this possible and can it be done with GUI Cconfiguration, I configured such way dids before only with IP Auth so with diggest auth its not working

Posts: 2

Participants: 2

Read full topic

@PetersPBX wrote:

Hello dear pbx fans
I very new so be a little patient
I installed FreePBX on RPI 3b+ with the image file
Did the raspbx-upgrade >> FreePBX 14.0.2.10 ‘VoIP Server’
Can login to the UI
Now i’am stuck
Login with new user and password
-I do not get the register new pbx
-I can not start the wizard again, it fails in the menu

If i make a new admin user with access to all items i get a warning

NOTE: Authorization Type is set to 'usermanager' in Advanced Settings - note that this module is not currently providing full access control and is only used as a failover, stop-gap until this pane is fully migrated to User Manager. You will still be able to login with the users below as long as their username does not exist in User Manager

I reinstalled everything but i’am stuck at the same point
Where can or should i change it , i can not find " Authorization Type is set to ‘usermanager’ in Advanced Settings " in the advanced menu

Posts: 1

Participants: 1

Read full topic

@lblokland wrote:

Hi,

I’m trying to build a new dialplan to redirect a call to an extension based on the dialed number (DID) but I already stuck at the very beginning.
I’ve created a custom dialplan in the ‘extensions_custom.conf’ and added the following lines;

[dialplan-did-to-extention]
exten => _X.,1,NoOp(entering dialplan-did-to-extention ${CALLERID(DNID)})
exten => _X.,n(BUSY),hangup

In Freepbx I rerouted the inbound did to a custom application, that has as a target ‘dialplan-did-to-extention,${EXTEN},1’

I was expecting a log entry starting with ‘‘entering dialplan-did-to-extention’’ and ending with the hangup, but instead the call is being redirected to the callerid (I’m dailing with my cell phone, so I end up with my own vm welcome message)]

The log shows:
Goto (dialplan-did-to-extention,##my-did-nr###,1)
– Executing [##my-did-nr###@dialplan-did-to-extention:1] NoOp(“SIP/my-trunk-name-0000073f”, “entering dialplan-did-to-extention ##my-did-nr###”) in new stack
– Executing [##my-did-nr###@dialplan-did-to-extention:2] Hangup(“SIP/my-trunk-name-0000073f”, “”) in new stack
== Spawn extension (dialplan-did-to-extention, ##my-did-nr###, 2) exited non-zero on ‘SIP/my-trunk-name-0000073f’

Can anyone explain why that call is being transfered to my own number?

Thanks
Leon

Posts: 1

Participants: 1

Read full topic

@hunterman wrote:

Hello everyone
I need to change the status of “FreePBX Statistics” on Dashboard Asterisk from Hour to Minutes or seconds, because I need to know and see who extensions on the live line at that moment?
OR how can i know who extension be call " from extension To extensions agent ", Any method can i know that?
THANKS

Posts: 1

Participants: 1

Read full topic

@bbrotschi wrote:

I am new to Asterisk and FreePBX, but follow directions pretty well. I was following the instructions at the following site
https://wiki.freepbx.org/plugins/servlet/mobile?contentId=4161715#FreePBXDistroFirstStepsAfterInstallation-ConfiguringaPhoneUsingEndPointManager(EPM)

to conduct the initial setup of my PBX
PBX Firmware:12.7.5-1902-3.sng7
PBX Service Pack:1.0.0.0
FreePBX 14.0.5.25

and made it as far as “Configuring a Phone Using EndPoint Manager (EPM)”, I purchased a desktop phone (model A20 from Asterisk) and also purchased the fully functional version of EPM. What I can not seem to locate is the correct EPM template for this phone set. When I conduct the “Network Scan” function from within EPM. The phone is identified as a Dignum brand, available templates only include “dignum_default” and model D40 - D80

Posts: 7

Participants: 4

Read full topic

@ashcortech wrote:

After some module updates on a FreePBX13 system, Pagin and intercomm are no longer working properly. When dialing the intercom group the phones ring instead of beep and go offhook.

I’ve now seen this on two systems. The first I couldn’t find a cause so I just re-loaded it from scratch. Now I have a second with the same isssue so I’d like to figure out what’s causing it.

I tried rolling back the paging and intercomm module but that didn’t fix it.

are there any current known issues?

Posts: 2

Participants: 1

Read full topic

@GSnover wrote:

Ok - I have to use the following macro with my provider to apply a FROM: header on my outbound calls to set my own CallerID

[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp(Adding FROM: Header for Provider)
exten => s,n,Set(CallerIDString=${CALLERID(all)})
exten => s,n,SipAddHeader(From: $CallerIDString)
exten => s,n,MacroExit()

But I have a company that is actually TWO companies - they use DISA to control when somebody from company A wants to look like someone from company B.

I think I am referencing the wrong variable here: ${CALLERID(all)}. That variable just picks up the Extension’s Caller-ID.

Can anyone tell me what the correct variable I should be looking for is?

Thanks!

Greg

Posts: 1

Participants: 1

Read full topic

@sentinelace wrote:

I need to send out a different trunk for fax only. I have an ATA registered as extension 600. How do I force it to go out a specific trunk?

Posts: 3

Participants: 2

Read full topic

@pbx3 wrote:

Hello,

I am running FreePBX 14.0.5.25. Under the user management module, there is a search window on the right side. I mistakenly put in a username, not one already registered with the user management module.

My problem is that I cannot remove this username from the search window. Every time I clear it it immediately reappears.

This makes it impossible to see any other content on the page.

I am wondering if other folks are having this issue, and/or if it is a bug I should report.

Thanks

Posts: 2

Participants: 2

Read full topic

@EdCazini wrote:

Hi. I have a successful setup, internal conversations are fine, and some outbound external (via sip trunk) calls as well, but not all. Inbound is fine too. For some outbound trunk calls to specific external destinations, the internal caller can’t hear the remote side’s audio for about 10-20 seconds although the remote can hear the caller. My setup is a CentOS 6.10, FPBX-13.0.192.19, Asterisk 13.23.0.

Here’s a link to the network diagram.
drive[dot]google[dot]com/file/d/1oFFO6T3fIScmhMAXT6zwiCaHxP52Xj2Q/view

Also here are some settings:

SIP:

[general]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
allow=g722
allow=h264
allow=mpeg4
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
match_auth_username=yes
accept_outofcall_message=yes
outofcall_message_context=astsms
rtpend=20000
rtpstart=10000
context=from-sip-external
callevents=yes
tcpenable=no
bindport=5060
jbenable=no
tlsbindaddr=[::]:5060
notifyhold=yes
tlsclientmethod=sslv2
tlsenable=no
srvlookup=yes
allowguest=no
defaultexpiry=120
minexpiry=60
rtptimeout=30
g726nonstandard=no
videosupport=yes
maxcallbitrate=384
canreinvite=no
rtpholdtimeout=300
rtpkeepalive=1
checkmwi=10
notifyringing=yes
registertimeout=20
maxexpiry=3600
registerattempts=0
nat=force_rport,comedia
ALLOW_SIP_ANON=no
callerid=Unknown
externip=d.d.d.d
localnet=a.a.a.a/22
;I have to set the SBC IP address as localnet otherwise all internal-to-external (vice-versa) calls can’t hear each other
localnet=c.c.c.c/32
language=en

Typical extension:

[XXXX]
deny=0.0.0.0/0.0.0.0
secret=XXXX
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=yes ;other settings have no effect
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/XXXX
permit=0.0.0.0/0.0.0.0
callerid=whatever
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

SIP trunk:

[SIPTRUNK-1]
disallow=all
username=ZZZZZZZZZZ
; provider does not require secret
type=peer
qualify=yes
nat=force_rport,comedia
insecure=invite,port
host=c.c.c.c
dtmfmode=rfc2833
context=from-trunk
allow=ulaw
allow=alaw
allow=gsm

What’s weird is that when I did a TCPDump on my server, in Wireshark --> Telephony --> VoIP Calls --> Play Streams, the remote side’s audio is there. It just did not arrive at the caller.

Here is also the link to the PCAP file.
drive[dot]google[dot]com/open?id=17QMZcFL933gfDytp3x9vRk8wGXJNiv4k

Sorry for the links, I just registered, I can’t post URL links nor attach files.
Any help appreciated. I’m prepared to pay if necessary. TIA. Edwin

Posts: 1

Participants: 1

Read full topic

@steve_pbuk wrote:

I have a bunch of Sangoma S500 connected to FreePBX via the built in VPN Server. Everything works fine for making and receiving calls, etc.

I also setup a user within FreePBX for the purpose of using the VPN profile downloaded from UCP to connect my laptop over the VPN for administrative tasks. This works fine and I can connect to the FreePBX server via 10.8.0.1.

I also want to be able to access the web GUI for the Sangoma phones that are connected via VPN. This is where I run into problems. From my laptop I can only ping the FreePBX server on 10.8.0.1. If I try to ping the 10.8.x.x of the Sangoma phones it fails and I can not access the GUI.

I’ve checked and I have ticked to enable the 10.8.0.0 route under VPN Server Settings:

image.png1336×504 19.9 KB

Am I missing something? Is this something that should work?

Posts: 1

Participants: 1

Read full topic

@pixa241 wrote:

So, we started running into this issue a few weeks ago. An outbound call is being made, and it immediately hangs up or says call failure, yet the number we call calls back and say we called but immediately hanged up. We are using FreePBX 14 with all modules up to date. We are using Grandstream GXP2140 phones also on the latest firmware. Our SIP trunk provider doesn’t see any issue on their end.

https://pastebin.com/q6swLjyj

Posts: 1

Participants: 1

Read full topic

@ozarktech wrote:

UCP daemon is not running in the dashboard. And when logged into the UCP and click save to apply changes it says You are currently working in offline mode and the changes don’t save. Restarted pbx and no change. This is on PBXact 13.0.195.26 and firmware is 10.13.66-21. All modules are up to date.

Posts: 3

Participants: 2

Read full topic

@laca wrote:

Hello,

I started to have issues with incoming SIP calls. My provider is sending the invites but my sip server is not responding on the invites any more. Out going call are working just fine.
On the the server with sip debugging on I don’t see any incoming messages from the provider.
Did someone experienced the same issue? Can someone point me in the right direction of the root cause?

sip invite.jpg709×268 47.3 KB

Kind regards,
Laurent

Posts: 2

Participants: 2

Read full topic

@lolopeople wrote:

I am Having problem with setting the correct permission for user to add contacts when they are in UCP.
users are able to create their own group and contacts within the group they created but users are unable to create groups they are in.
Example:
User X , is in the Support Group, while they are able to view all the contacts within the support Group but they are not able to add new contacts.
Did I set the wrong permission for the users? or what are the regular way for users to use UCP contact.
My hope is that Users in X department can add their own contacts within their group, so that whenever the callers called , callers’ Display Name is showed on the phone.
Thanks

Posts: 1

Participants: 1

Read full topic

@sentinelace wrote:

I need a solution to showing out of office on the blf key. Meaning, if they hit DND, instead of showing red on the BLF, change state to off. I am using yealink. Is this possible or a similar solution?

Posts: 4

Participants: 2

Read full topic

@aspieboy77 wrote:

Hi all I’ve currently transitioned to aws or amazon web services for hosting my freepbx server but i’m having this one issue when i call out there is no sound on either end plus i’m unable to call into the freepbx server on the main phone line i get a busy tone. thanks for all the help you guys can give in advance

2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/251-00000004”, “TOUCH_MONITOR=1551024221.4”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/251-00000004”, “AMPUSER=251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/251-00000004”, “0?report”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/251-00000004”, “1?Set(REALCALLERIDNUM=251)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/251-00000004”, “AMPUSER=251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/251-00000004”, “0?limit”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/251-00000004”, “AMPUSERCIDNAME=Christopher Miguez”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“SIP/251-00000004”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“SIP/251-00000004”, “0?report”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:10] Set(“SIP/251-00000004”, “AMPUSERCID=251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:11] Set(“SIP/251-00000004”, “__DIAL_OPTIONS=Ttr”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:12] Set(“SIP/251-00000004”, “CALLERID(all)=“Christopher Miguez” <251>”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:13] GotoIf(“SIP/251-00000004”, “0?limit”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:14] ExecIf(“SIP/251-00000004”, “1?Set(GROUP(concurrency_limit)=251)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:15] ExecIf(“SIP/251-00000004”, “0?Set(CHANNEL(language)=)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:16] NoOp(“SIP/251-00000004”, “Macro Depth is 1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“SIP/251-00000004”, “1?report2:macroerror”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“SIP/251-00000004”, “1?continue”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/251-00000004”, “CALLERID(number)=251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:38] Set(“SIP/251-00000004”, “CALLERID(name)=Christopher Miguez”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“SIP/251-00000004”, “0?cnum”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:40] Set(“SIP/251-00000004”, “CDR(cnam)=Christopher Miguez”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:41] Set(“SIP/251-00000004”, “CDR(cnum)=251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-user-callerid:42] Set(“SIP/251-00000004”, “CHANNEL(language)=en”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [96823006910@from-internal:2] Gosub(“SIP/251-00000004”, “sub-record-check,s,1(out,96823006910,force)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:1] GotoIf(“SIP/251-00000004”, “0?initialized”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:2] Set(“SIP/251-00000004”, “__REC_STATUS=INITIALIZED”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:3] Set(“SIP/251-00000004”, “NOW=1551024221”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:4] Set(“SIP/251-00000004”, “__DAY=24”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:5] Set(“SIP/251-00000004”, “__MONTH=02”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:6] Set(“SIP/251-00000004”, “__YEAR=2019”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:7] Set(“SIP/251-00000004”, “__TIMESTR=20190224-160341”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:8] Set(“SIP/251-00000004”, “__FROMEXTEN=251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:9] Set(“SIP/251-00000004”, “__MON_FMT=wav”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:10] NoOp(“SIP/251-00000004”, “Recordings initialized”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:11] ExecIf(“SIP/251-00000004”, “0?Set(ARG3=dontcare)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:12] Set(“SIP/251-00000004”, “REC_POLICY_MODE_SAVE=”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:13] ExecIf(“SIP/251-00000004”, “0?Set(REC_STATUS=NO)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:14] GotoIf(“SIP/251-00000004”, “3?checkaction”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (sub-record-check,s,17)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@sub-record-check:17] GotoIf(“SIP/251-00000004”, “1?sub-record-check,out,1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (sub-record-check,out,1)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [out@sub-record-check:1] NoOp(“SIP/251-00000004”, “Outbound Recording Check from 251 to 96823006910”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [out@sub-record-check:2] Set(“SIP/251-00000004”, “RECMODE=force”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [out@sub-record-check:3] ExecIf(“SIP/251-00000004”, “0?Goto(routewins)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [out@sub-record-check:4] ExecIf(“SIP/251-00000004”, “1?Goto(routewins)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (sub-record-check,out,7)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [out@sub-record-check:7] Gosub(“SIP/251-00000004”, “recordcheck,1(force,out,96823006910)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“SIP/251-00000004”, “Starting recording check against force”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“SIP/251-00000004”, “force”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (sub-record-check,recordcheck,5)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:5] Set(“SIP/251-00000004”, “__REC_POLICY_MODE=FORCE”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:6] GotoIf(“SIP/251-00000004”, “1?startrec”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (sub-record-check,recordcheck,16)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:16] NoOp(“SIP/251-00000004”, “Starting recording: out, 96823006910”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:17] Set(“SIP/251-00000004”, “__CALLFILENAME=out-96823006910-251-20190224-160341-1551024221.4”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:18] MixMonitor(“SIP/251-00000004”, “2019/02/24/out-96823006910-251-20190224-160341-1551024221.4.wav,abi(LOCAL_MIXMON_ID),”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:19] Set(“SIP/251-00000004”, “__MIXMON_ID=0x7f3a58057ee0”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:20] Set(“SIP/251-00000004”, “__RECORD_ID=SIP/251-00000004”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:21] Set(“SIP/251-00000004”, “__REC_STATUS=RECORDING”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:22] Set(“SIP/251-00000004”, “CDR(recordingfile)=out-96823006910-251-20190224-160341-1551024221.4.wav”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:23] Return(“SIP/251-00000004”, “”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [out@sub-record-check:8] Return(“SIP/251-00000004”, “”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [96823006910@from-internal:3] ExecIf(“SIP/251-00000004”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [96823006910@from-internal:4] Set(“SIP/251-00000004”, “MOHCLASS=default”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [96823006910@from-internal:5] ExecIf(“SIP/251-00000004”, “0?Set(TRUNKCIDOVERRIDE=16822266684)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [96823006910@from-internal:6] Set(“SIP/251-00000004”, “_NODEST=”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [96823006910@from-internal:7] Macro(“SIP/251-00000004”, “dialout-trunk,1,16823006910,off”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:1] Set(“SIP/251-00000004”, “DIAL_TRUNK=1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf(“SIP/251-00000004”, “0?Set(DIAL_OPTIONS=tr)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf(“SIP/251-00000004”, “0?sub-pincheck,s,1()”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:4] ExecIf(“SIP/251-00000004”, “0?Set(CALLERID(num)=251)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:5] GotoIf(“SIP/251-00000004”, “0?disabletrunk,1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:6] Set(“SIP/251-00000004”, “DIAL_NUMBER=16823006910”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:7] Set(“SIP/251-00000004”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:8] Set(“SIP/251-00000004”, “OUTBOUND_GROUP=OUT_1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:9] Set(“SIP/251-00000004”, “DIAL_TRUNK_OPTIONS=T”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/251-00000004”, “0?nomax”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf(“SIP/251-00000004”, “0?chanfull”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:12] GotoIf(“SIP/251-00000004”, “0?skipoutcid”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:13] Macro(“SIP/251-00000004”, “outbound-callerid,1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:1] NoOp(“SIP/251-00000004”, “251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:2] NoOp(“SIP/251-00000004”, “”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:3] NoOp(“SIP/251-00000004”, “all”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf(“SIP/251-00000004”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf(“SIP/251-00000004”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:6] ExecIf(“SIP/251-00000004”, “0?Set(REALCALLERIDNUM=251)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:7] ExecIf(“SIP/251-00000004”, “0?Set(AMPUSER=251)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:8] GotoIf(“SIP/251-00000004”, “1?normcid”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (macro-outbound-callerid,s,12)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:12] Set(“SIP/251-00000004”, “USEROUTCID=Christopher Miguez”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:13] Set(“SIP/251-00000004”, “EMERGENCYCID=”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:14] Set(“SIP/251-00000004”, “TRUNKOUTCID=16822266684”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:15] GotoIf(“SIP/251-00000004”, “1?trunkcid”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (macro-outbound-callerid,s,20)
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:20] ExecIf(“SIP/251-00000004”, “1?Set(CALLERID(all)=16822266684)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf(“SIP/251-00000004”, “1?Set(CALLERID(all)=Christopher Miguez)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf(“SIP/251-00000004”, “1?Set(CALLERID(all)=16822266684)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf(“SIP/251-00000004”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/251-00000004”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:25] Set(“SIP/251-00000004”, “CDR(outbound_cnum)=16822266684”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:26] Set(“SIP/251-00000004”, “CDR(outbound_cnam)=”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:14] GosubIf(“SIP/251-00000004”, “0?sub-flp-1,s,1()”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:15] Set(“SIP/251-00000004”, “OUTNUM=16823006910”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:16] Set(“SIP/251-00000004”, “custom=SIP/gotrunk”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf(“SIP/251-00000004”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/251-00000004”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:19] Macro(“SIP/251-00000004”, “dialout-trunk-predial-hook,”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/251-00000004”, “”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:20] GotoIf(“SIP/251-00000004”, “0?skipcrm”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:21] Set(“SIP/251-00000004”, “__CRM_DIRECTION=OUTBOUND”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:22] Set(“SIP/251-00000004”, “__CRM_DESTINATION=16823006910”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:23] Set(“SIP/251-00000004”, “__CRM_SOURCE=251”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:24] AGI(“SIP/251-00000004”, “sangomacrm.agi”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-02-24 16:03:41] VERBOSE[3328][C-00000002] app_mixmonitor.c: Begin MixMonitor Recording SIP/251-00000004
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] res_agi.c: <SIP/251-00000004>AGI Script sangomacrm.agi completed, returning 0
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:25] Set(“SIP/251-00000004”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp(“SIP/251-00000004”, “CRM Finished”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf(“SIP/251-00000004”, “0?bypass,1”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf(“SIP/251-00000004”, “1?Set(CONNECTEDLINE(num,i)=16823006910)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf(“SIP/251-00000004”, “1?Set(CONNECTEDLINE(name,i)=CID:16822266684)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf(“SIP/251-00000004”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)16822266684)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf(“SIP/251-00000004”, “0?customtrunk”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:32] Dial(“SIP/251-00000004”, “SIP/gotrunk/16823006910,300,Tb(func-apply-sipheaders^s^1,(1))”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] netsock2.c: Using SIP RTP TOS bits 184
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] netsock2.c: Using SIP RTP CoS mark 5
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] app_stack.c: SIP/gotrunk-00000005 Internal Gosub(func-apply-sipheaders,s,1(1)) start
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/gotrunk-00000005”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“SIP/gotrunk-00000005”, “Applying SIP Headers to channel SIP/gotrunk-00000005”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/gotrunk-00000005”, “TECH=SIP”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:4] Set(“SIP/gotrunk-00000005”, “SIPHEADERKEYS=”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:5] While(“SIP/gotrunk-00000005”, “0”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] app_while.c: Jumping to priority 12
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:13] Return(“SIP/gotrunk-00000005”, “”) in new stack
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] app_stack.c: Spawn extension (from-trunk, 96823006910, 1) exited non-zero on ‘SIP/gotrunk-00000005’
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] app_stack.c: SIP/gotrunk-00000005 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2019-02-24 16:03:41] VERBOSE[3326][C-00000002] app_dial.c: Called SIP/gotrunk/16823006910
[2019-02-24 16:03:43] VERBOSE[3326][C-00000002] app_dial.c: SIP/gotrunk-00000005 is making progress passing it to SIP/251-00000004
[2019-02-24 16:03:44] VERBOSE[3326][C-00000002] app_dial.c: SIP/gotrunk-00000005 is ringing
[2019-02-24 16:03:44] WARNING[3326][C-00000002] translate.c: no samples for ulawtolin
[2019-02-24 16:03:46] VERBOSE[3326][C-00000002] app_dial.c: SIP/gotrunk-00000005 requested media update control 26, passing it to SIP/251-00000004
[2019-02-24 16:03:46] VERBOSE[3326][C-00000002] app_dial.c: SIP/gotrunk-00000005 answered SIP/251-00000004
[2019-02-24 16:03:46] VERBOSE[3346][C-00000002] bridge_channel.c: Channel SIP/gotrunk-00000005 joined ‘simple_bridge’ basic-bridge <51778390-6941-4370-a157-2c110638e617>
[2019-02-24 16:03:46] VERBOSE[3326][C-00000002] bridge_channel.c: Channel SIP/251-00000004 joined ‘simple_bridge’ basic-bridge <51778390-6941-4370-a157-2c110638e617>
[2019-02-24 16:03:52] WARNING[2517] chan_sip.c: Retransmission timeout reached on transmission 90f168d2-87ce0e8d@192.168.1.185 for seqno 102 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2019-02-24 16:03:52] WARNING[2517] chan_sip.c: Hanging up call 90f168d2-87ce0e8d@192.168.1.185 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] bridge_channel.c: Channel SIP/251-00000004 left ‘simple_bridge’ basic-bridge <51778390-6941-4370-a157-2c110638e617>
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] app_macro.c: Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on ‘SIP/251-00000004’ in macro ‘dialout-trunk’
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] pbx.c: Spawn extension (from-internal, 96823006910, 7) exited non-zero on ‘SIP/251-00000004’
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] pbx.c: Executing [h@from-internal:1] Macro(“SIP/251-00000004”, “hangupcall”) in new stack
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/251-00000004”, “1?theend”) in new stack
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/251-00000004”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/251-00000004”, “SIP/gotrunk-00000005 monior file= /var/spool/asterisk/monitor/2019/02/24/out-96823006910-251-20190224-160341-1551024221.4.wav”) in new stack
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-hangupcall:5] AGI(“SIP/251-00000004”, “attendedtransfer-rec-restart.php,SIP/gotrunk-00000005,/var/spool/asterisk/monitor/2019/02/24/out-96823006910-251-20190224-160341-1551024221.4.wav”) in new stack
[2019-02-24 16:03:52] VERBOSE[3326][C-00000002] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2019-02-24 16:03:52] VERBOSE[3346][C-00000002] bridge_channel.c: Channel SIP/gotrunk-00000005 left ‘simple_bridge’ basic-bridge <51778390-6941-4370-a157-2c110638e617>
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] res_agi.c: <SIP/251-00000004>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“SIP/251-00000004”, “”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/251-00000004’ in macro ‘hangupcall’
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/251-00000004’
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] app_stack.c: SIP/251-00000004 Internal Gosub(crm-hangup,s,1) start
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/251-00000004”, “Sending Hangup to CRM”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/251-00000004”, “HANGUP CAUSE: 18”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/251-00000004”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/251-00000004”, “MASTER CHANNEL: 1551024221.4 = 1551024221.4”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/251-00000004”, “0?return”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/251-00000004”, “__CRM_HANGUP=1”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/251-00000004”, “sangomacrm.agi”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] res_agi.c: <SIP/251-00000004>AGI Script sangomacrm.agi completed, returning 0
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/251-00000004”, “”) in new stack
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/251-00000004’
[2019-02-24 16:03:53] VERBOSE[3326][C-00000002] app_stack.c: SIP/251-00000004 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2019-02-24 16:03:53] VERBOSE[3328][C-00000002] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2019-02-24 16:03:53] VERBOSE[3328][C-00000002] app_mixmonitor.c: End MixMonitor Recording SIP/251-00000004

Posts: 3

Participants: 3

Read full topic