@antonis77 wrote:
Hi,
Is there any module for FreePBX to see active calls on the system, incoming or outgoing.
In a way i want to check which extension is busy or not through o gui.I think FOP2 does the job, but i was wondering if FreePBX has a module
Posts: 1
Participants: 1
@santorio wrote:
I have installed Asterisk 13 from source and then install all example configuration files with command “make config”. Then I installed FreePBX 13 from source and I suspect that FreePBX doesn’t write all configs over example files. Because I had to write AMI login and password to manager.conf by hand.
How can I recreate all files that must be installed through normal installation process?
Posts: 3
Participants: 3
@RalphGraham wrote:
I have speed dial set up and working *0 and a number which works fine for UK calls. I added 2 international numbers UK dial code is 00 for international followed by country code and number. These do not work and appear blocked. I found a reference to enable pinless dialling which I enabled but still not working.
There appears to be a setting somewhere I need to change.
Any help please.
Posts: 2
Participants: 2
@TheBlueKingLP wrote:
Hello, I am trying to add a custom header in the REGISTER packets, but I only found posts about adding it when someone dials, is there a way to achieve this?
Posts: 1
Participants: 1
@jfinstrom wrote:
FOr those who missed the update from matt…
Asterisk 13 is now security fix only and people should migrate to a 16/17
https://wiki.asterisk.org/wiki/display/AST/Asterisk+VersionsWith Asterisk 18 chan_sip is disabled by default. People should work on migrating all sip stuff to pjsip
Roadmap for FreePBX:
Upgrade distro to an EL8 base
FreePBX 16 focus on PHP 7.X support.
Posts: 1
Participants: 1
@hmg21215 wrote:
When I try to run fwconsole ma install core, i get the following error: [Exception]
Unknown Error Code 42000, Message SQLSTATE[42000]: Syntax error or access
violation: 1071 Specified key was too long; max key length is 767 bytesI tried to install core because the system would not reload, with an error of 255 but no explanation. There was also a cronmanager error. I yum updated, and got more errors and was advised by the system to install the core through fwconsole. My core and framework grep is:
| core | 14.0.1.25 | Disabled; Pending upgrade to 13.0.131.28 | GPLv3+ |
| framework | 13.0.197.28 | EnabledThe system is not working (it has been active for a couple of years. No calls, no web GUI, just putty and Winscp.
Posts: 3
Participants: 1
@vespino wrote:
I have a CID lookup script in place which looks up a number for caller id and also looks up if the number is on a spam list (telemarketing, etc). This script written in PHP works great, it shows me if I should not pick up and if not also adds the number to my blacklist. The next thing I was hoping to do is drop the call instead of just warning me. Can this be done from a PHP script? The script is running locally so I have access to the resources of the server.
This looks like it could work, but I don’t see how I can run this from my PHP script: https://confluence.panio.info/display/PUBL/HOW+to+hang+up+an+active+or+hung+call+in+FreePBX+or+Asterisk
Posts: 1
Participants: 1
@longqvo wrote:
I am looking for some guidance here. Our district recently purchased a softphone from ClearlyIP but we didn’t get a 1 to 1 license. We are using Active Directory sync to create extension and User Manager accounts. If the user is part of “Freepbx Users” Groups and ipphone is filled out, freepbx will create the extension and user manager. Which has been working fine.
My thinking here, I will create a 2nd group called “ucp enabled” in AD. Which will create a 2nd Group in Freepbx so I can enable softphones for those users in that group. This does create the 2nd group in FreePBX for me but it also creates a 2nd user in User manager for each user in “ucp enabled” group. What am I doing wrong here? Below are the setting from my enable UCP Setting, (I am doing this on a test server so I picked UCP as a setting test). Thanks in advance
Posts: 2
Participants: 1
@JoshuaCadman wrote:
I have a hosted extension service with Dial9 as I am unable to use the SIP service they offer as it requires a static IP. For months, I have set this up in FreePBX by setting the DID to any so that calls to the phone number would work correctly.
I now want to add a new number from Dial9 and this means that I need to sort out an issue I have put off for months with the DID.
The register string for the chan_sip is correct with the username:password@domain/username and under cdr the did is working correctly with the extension username ‘joshuacadman2-kittiwakeroad’. The only issue I have is editing the inbound route to ‘_joshuacadman2-kittiwakeroad’ this then says that the number is incorrect.
I know the issue will be the dash in the DID however I am only able to change the last part of the username after the dash.
Has anyone had this issue before as I would understand if no DID was detected
Posts: 2
Participants: 2
@robfantini wrote:
this fails on one of our systems, works OK on the main phone system:
yum update --skip-broken Transaction check error: file /usr/lib64/python3.6/ensurepip/__init__.py from install of python3-libs-3.6.8-13.el7.x86_64 conflicts with file from package python36u-libs-3.6.7-1.ius.centos7.x86_64 file /usr/lib64/python3.6/test/support/__init__.py from install of python3-libs-3.6.8-13.el7.x86_64 conflicts with file from package python36u-libs-3.6.7-1.ius.centos7.x86_64any clues on a path to solve the error?
Posts: 1
Participants: 1
@unison wrote:
We have moved all our trunks from chan_sip to pjsip - mostly without any issues.
We have one provider that requires that the from address in our SIP invite is our user… which is achieved using ‘From User’ option in the pjsip advanced settings…
However, when using chan_sip any the SIP From Display Name was maintained - so when forwarding a call from a queue, we could prefix this with Queue: Callers Name - which would intern make it through to the 3rd party provider.
From a packet capture; using chan_sip, it looks like (my user is 123456789)
From: "Queue: First Last" <sip:123456789@my.server.ip.add>Using PJSIP it looks like:
From: <sip:123456789@my.server.ip.add>Anyway to get the displayname back in to the PJSIP invite?
Posts: 1
Participants: 1
@duncanidaho wrote:
I am getting this error. I can ping mirror.freepbx.org and have tried fwconsole setting MODULE_REPO https://mirror.freepbx.org and still get the error. Is there something else I should be doing? I have rebooted machine, tried upgradeall, and have rebooted my router. Any help would be appreciated. My system is as follows:
PBX Version: 15.0.16.75
PBX Distro: 12.7.8-2008-1.sng7
Asterisk Version: 13.36.0
Warning: Cannot connect to online repository(s) (https://mirror.freepbx.org). Online modules are not available.
Posts: 1
Participants: 1
@jklin wrote:
My situation is as follows - and this may not be uncommon: my phone provider does not offer a SIP trunk, but individual SIP accounts for the series of numbers we have (e.g. 5553201, 5553202, 5553203, …).
Each colleague gets assigned on of these public numbers as “their” respective direct line (i.e. incoming calls on that number should primarily ring on that phone and outgoing calls from that phone shall go through that trunk). One number will be used as the front-desk/receptionist.I would need to setup multiple extensions - one for each colleague to connect their phone.
What is the best strategy for organizing this?
Do I need multiple inbound and outbound routes, one for each extension-trunk combo or is there a way to handle this better, e.g. if extensions and trunks are named in a particular fashion?
Posts: 2
Participants: 2
@GSnover wrote:
Ok - We are doing a Law Day Call-In for our State Bar this morning, but with Covid, instead of having the agents on attached phones, all the members of the Queues are on Cel Phones remotely.
I think the problem is that Asterisk is not detecting that an Agent is already on a call from another queue and is therefore offering the caller from the other Queue - here is the setup:
Queue A:
Agent A
Agent B
Agent CQueue B:
Agent A
Agent D
Agent ESo agent A is in both Queue A and Queue B (Their Cel Phone) - problem seems to be that when they are on a call for Queue A and a call is offered for Queue B, it goes ahead and offers the call from Queue B - Even though I have set both Queues to Skip Busy on the ring strategy.
I don’t think there is a way around this - has anyone else seen this behavior?
Posts: 13
Participants: 4
@openpbx wrote:
When I select to send an email to a user the LAN IP is listed for the UCP and password reset link. Where can I set it to be the public IP address or even better a web domain?
Posts: 3
Participants: 2
@sorvani wrote:
It started with weirdness when sysadmin was installed.
Stage 3 Download and Install Sysadmin Detected Missing Dependency of: backup 15.0.8.88 Downloading Missing Dependency of: backup 15.0.8.88 Module backup successfully downloaded Installing Missing Dependency of: backup 15.0.8.88 The following error(s) occured: - The Module Named "filestore" is required. Whoops\Exception\ErrorException: Class 'ComposerAutoloaderInitc5668f088e1d3d1331eec9932c333408' not found in file /var/www/html/admin/modules/backup/vendor/autoload.php on line 7 Stack trace: 1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/backup/vendor/autoload.php:7But things seemed to move on. And filestore was later installed as a dependency of backup.
/me looks at prior message again…
/me shakes head.Download and Install backup Detected Missing Dependency of: filestore Downloading Missing Dependency of: filestore Module filestore successfully downloaded Installing Missing Dependency of: filestore Generating CSS... Done Installed Missing Dependency of: filestore Migrating legacy backupjobs Moving servers to filestore Unable to map 'Config server' of type 'mysql Unable to map 'CDR server' of type 'mysql Migrating legacy backups to the new backup Cleaning up old data Removing table backup. Removing table backup_cache. Removing table backup_details. Removing table backup_items. Removing table backup_server_details. Removing table backup_servers. Removing table backup_template_details. Removing table backup_templates. Generating CSS...DoneSo the upgrade was moving along… Then it tried to install endpoint.
Download and Install endpoint Detected Missing Dependency of: sysadmin 14.0.11 Found local Dependency of: sysadmin 15.0.16.13 Installing Missing Dependency of: sysadmin 14.0.11 Updating tables sysadmin_options, sysadmin_update_log, sysadmin_fail2ban... Done Sangoma PnP Server activated Generating CSS... Done Installed Missing Dependency of: sysadmin 14.0.11 Detected Missing Dependency of: paging 15.0.4.18 Downloading Missing Dependency of: paging 15.0.4.18 Module paging successfully downloaded Installing Missing Dependency of: paging 15.0.4.18 Updating tables paging_groups, paging_autoanswer, paging_config, paging_core_routing... Done Generating CSS...Done Installed Missing Dependency of: paging 15.0.4.18 Create symlink...Done Checking database tables... Done Migrating tables as required... Done Checking Settings and Defaults... Done Generating Configs...Done Downloading Firmware... Done (Background) Generating CSS... DoneEventually it completed and I reloaded.
No Modules left to upgrade. The PBX has successfully upgraded [root@pbx ~]# sudo fwconsole reload Reload Started Reload CompleteThen check the online version of sysadmin and endpoint. They seem up to date.
[root@pbx ~]# fwconsole ma listonline | grep "sysadmin\|endpoint" | endpoint | 15.0.27.43 | Enabled and up to date | Commercial | | sysadmin | 15.0.16.13 | Enabled and up to date | Commercial |And yes, EPM is licensed for updates.
So it seems to be some weird messaging only.
Posts: 1
Participants: 1
@dancarter1220 wrote:
Hi,
I have a PBXACT 60.
Current PBX Version:14.0.13.28
Current System Version:
12.7.6-2002-2.sng7
I have an issue with call recordings.
If a call (Inbound or outbound) is made for example to or from ext 200, the call is recorded. If that call is then transferred to ext 201 (If attended transfer the call between 200 and 201 is still recorded) the recording stops.
The call itself is logged in CDR along with call duration but the recording is not available there isn’t one.I have forced recording on the extensions and also inbound on the Queue and outbound on the routes.
Has anybody experienced this and know of a fix?
Posts: 1
Participants: 1
@m0soo wrote:
Hi there, I seem to be having an issue dialling my parents (local) telephone number. I last called them on the 15th October and all was ok. Now every time I try and call them, I get a message that says I have dialled an invalid number. I can dial all other numbers ok but not theirs. I have tried calling them from my mobile and it connects ok?
I have checked with their phone provider (TalkTalk) to see if somehow my number has been blocked but it hasn’t. They can call me ok but I cant call them?
Any help you could give me would be much appreciated.
The number I am trying to call is 01723 35xxxx all other numbers usining the same area code and first digit (3) work ok!
I have tried dialling the number both with and without the area codes but it still fails.
Many thanks,
Kind regards
Jason
Posts: 2
Participants: 2
@ashleighkm wrote:
All of my external calls are disconnecting after 40 seconds or so. This is my sip trunk config
username=xxxxxxxxxxx
type=peer
setvar=T38GATEWAY=no
secret=xxxxxxxxxx
realm=zw.liquid.tel
qualify=yes
nat=yes
insecure=very
host=zw.liquid.tel
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=g722,alaw,ulawI have nat enabled between my firewall and the ipbx on ports 5060 and 14000-65000 for rtp.
i have enabled sip debug and this is what im getting, any help will be appreciated
<— SIP read from UDP:172.16.12.56:5060 —>
SIP/2.0 200 OK
From: “Unknown” sip:Unknown@172.16.80.210:5160;tag=as55c8ae9d
To: sip:zw.liquid.tel;tag=sip+1+31b40006+74caa44f
Via: SIP/2.0/UDP 172.16.80.210:5160;received=172.16.80.210;rport=29685;branch=z9hG4bK52330b18
Server: SIP/2.0
Max-Forwards: 70
Contact: sip:Unknown@172.16.80.210:5160
Call-ID: 78743925137a07f5477a3cc0611b1bfd@172.16.80.210:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.23(15.7.4)
Date: Sun, 25 Oct 2020 16:17:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘78743925137a07f5477a3cc0611b1bfd@172.16.80.210:5160’ Method: OPTIONS
== Setting global variable ‘SIPDOMAIN’ to ‘192.168.4.1’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Executing [167@from-internal:1] GotoIf(“PJSIP/192-0000013e”, “1?ext-local,167,1:followme-check,167,1”) in new stack
– Goto (ext-local,167,1)
– Executing [167@ext-local:1] Set(“PJSIP/192-0000013e”, “__RINGTIMER=15”) in new stack
– Executing [167@ext-local:2] Macro(“PJSIP/192-0000013e”, “exten-vm,167,167,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“PJSIP/192-0000013e”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/192-0000013e”, “TOUCH_MONITOR=1603642634.372”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/192-0000013e”, “AMPUSER=192”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/192-0000013e”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/192-0000013e”, “1?Set(REALCALLERIDNUM=192)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/192-0000013e”, “AMPUSER=192”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/192-0000013e”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“PJSIP/192-0000013e”, “AMPUSERCIDNAME=Gate 5”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/192-0000013e”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/192-0000013e”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/192-0000013e”, “AMPUSERCID=192”) in new stack
– Executing [s@macro-user-callerid:11] Set(“PJSIP/192-0000013e”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/192-0000013e”, “CALLERID(all)=“Gate 5” <192>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/192-0000013e”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/192-0000013e”, “0?Set(GROUP(concurrency_limit)=192)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“PJSIP/192-0000013e”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“PJSIP/192-0000013e”, “Macro Depth is 2”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/192-0000013e”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [s@macro-user-callerid:18] GotoIf(“PJSIP/192-0000013e”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:19] ExecIf(“PJSIP/192-0000013e”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
– Executing [s@macro-user-callerid:20] Set(“PJSIP/192-0000013e”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:21] GotoIf(“PJSIP/192-0000013e”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“PJSIP/192-0000013e”, “CALLERID(number)=192”) in new stack
– Executing [s@macro-user-callerid:38] Set(“PJSIP/192-0000013e”, “CALLERID(name)=Gate 5”) in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“PJSIP/192-0000013e”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“PJSIP/192-0000013e”, “CDR(cnam)=Gate 5”) in new stack
– Executing [s@macro-user-callerid:41] Set(“PJSIP/192-0000013e”, “CDR(cnum)=192”) in new stack
– Executing [s@macro-user-callerid:42] Set(“PJSIP/192-0000013e”, “CHANNEL(language)=en”) in new stack
– Executing [s@macro-exten-vm:2] Set(“PJSIP/192-0000013e”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“PJSIP/192-0000013e”, “__EXTTOCALL=167”) in new stack
– Executing [s@macro-exten-vm:4] Set(“PJSIP/192-0000013e”, “__PICKUPMARK=167”) in new stack
– Executing [s@macro-exten-vm:5] Set(“PJSIP/192-0000013e”, “RT=15”) in new stack
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:6] ExecIf(“PJSIP/192-0000013e”, “0?Macro(vm,167,DIRECTDIAL,)”) in new stack
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:7] ExecIf(“PJSIP/192-0000013e”, “0?MacroExit()”) in new stack
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:8] ExecIf(“PJSIP/192-0000013e”, “0?Gosub(ext-intercom,*80167,1())”) in new stack
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:9] ExecIf(“PJSIP/192-0000013e”, “0?MacroExit()”) in new stack
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:10] ExecIf(“PJSIP/192-0000013e”, “0?ChanSpy(PJSIP/167,q)”) in new stack
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:11] ExecIf(“PJSIP/192-0000013e”, “0?MacroExit()”) in new stack
[2020-10-25 18:17:14] WARNING[19364][C-000000bc]: chan_sip.c:23073 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:12] ExecIf(“PJSIP/192-0000013e”, “0?Macro(vm,167,DIRECTDIAL,)”) in new stack
– Executing [s@macro-exten-vm:13] ExecIf(“PJSIP/192-0000013e”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:14] ExecIf(“PJSIP/192-0000013e”, “0?Gosub(ext-intercom,*80167,1())”) in new stack
– Executing [s@macro-exten-vm:15] ExecIf(“PJSIP/192-0000013e”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:16] ExecIf(“PJSIP/192-0000013e”, “0?ChanSpy(PJSIP/167,q)”) in new stack
– Executing [s@macro-exten-vm:17] ExecIf(“PJSIP/192-0000013e”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:18] Gosub(“PJSIP/192-0000013e”, “sub-record-check,s,1(exten,167,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/192-0000013e”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/192-0000013e”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/192-0000013e”, “NOW=1603642634”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/192-0000013e”, “__DAY=25”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/192-0000013e”, “__MONTH=10”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/192-0000013e”, “__YEAR=2020”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/192-0000013e”, “__TIMESTR=20201025-181714”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/192-0000013e”, “__FROMEXTEN=192”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/192-0000013e”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/192-0000013e”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/192-0000013e”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/192-0000013e”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/192-0000013e”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/192-0000013e”, “5?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/192-0000013e”, “1?sub-record-check,exten,1”) in new stack
– Goto (sub-record-check,exten,1)
– Executing [exten@sub-record-check:1] NoOp(“PJSIP/192-0000013e”, “Exten Recording Check between 192 and 167”) in new stack
– Executing [exten@sub-record-check:2] Set(“PJSIP/192-0000013e”, “CALLTYPE=internal”) in new stack
– Executing [exten@sub-record-check:3] ExecIf(“PJSIP/192-0000013e”, “0?Set(CALLTYPE=)”) in new stack
– Executing [exten@sub-record-check:4] Set(“PJSIP/192-0000013e”, “CALLEE=dontcare”) in new stack
– Executing [exten@sub-record-check:5] ExecIf(“PJSIP/192-0000013e”, “0?Set(CALLEE=dontcare)”) in new stack
– Executing [exten@sub-record-check:6] GotoIf(“PJSIP/192-0000013e”, “0?callee”) in new stack
– Executing [exten@sub-record-check:7] GotoIf(“PJSIP/192-0000013e”, “1?caller”) in new stack
– Goto (sub-record-check,exten,13)
– Executing [exten@sub-record-check:13] Set(“PJSIP/192-0000013e”, “RECMODE=dontcare”) in new stack
– Executing [exten@sub-record-check:14] ExecIf(“PJSIP/192-0000013e”, “0?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:15] ExecIf(“PJSIP/192-0000013e”, “1?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:16] Gosub(“PJSIP/192-0000013e”, “recordcheck,1(dontcare,internal,167)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/192-0000013e”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/192-0000013e”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/192-0000013e”, “”) in new stack
– Executing [exten@sub-record-check:17] Return(“PJSIP/192-0000013e”, “”) in new stack
– Executing [s@macro-exten-vm:19] GotoIf(“PJSIP/192-0000013e”, “1?macrodial”) in new stack
– Goto (macro-exten-vm,s,25)
– Executing [s@macro-exten-vm:25] GosubIf(“PJSIP/192-0000013e”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:26] Macro(“PJSIP/192-0000013e”, “dial-one,15,HhTtr,167”) in new stack
– Executing [s@macro-dial-one:1] Set(“PJSIP/192-0000013e”, “DEXTEN=167”) in new stack
– Executing [s@macro-dial-one:2] Set(“PJSIP/192-0000013e”, “__CRM_SOURCE=192”) in new stack
– Executing [s@macro-dial-one:3] ExecIf(“PJSIP/192-0000013e”, “0?Set(__EXTTOCALL=167)”) in new stack
– Executing [s@macro-dial-one:4] Set(“PJSIP/192-0000013e”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:5] GosubIf(“PJSIP/192-0000013e”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:6] GosubIf(“PJSIP/192-0000013e”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:7] GotoIf(“PJSIP/192-0000013e”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,10)
– Executing [s@macro-dial-one:10] GotoIf(“PJSIP/192-0000013e”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:11] GotoIf(“PJSIP/192-0000013e”, “0?continue”) in new stack
– Executing [s@macro-dial-one:12] Set(“PJSIP/192-0000013e”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:13] GotoIf(“PJSIP/192-0000013e”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,25)
– Executing [s@macro-dial-one:25] GotoIf(“PJSIP/192-0000013e”, “0?next3:continue”) in new stack
– Goto (macro-dial-one,s,27)
– Executing [s@macro-dial-one:27] GotoIf(“PJSIP/192-0000013e”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GosubIf(“PJSIP/192-0000013e”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“PJSIP/192-0000013e”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“PJSIP/192-0000013e”, “DEVICES=167”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“PJSIP/192-0000013e”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“PJSIP/192-0000013e”, “0?Set(DEVICES=67)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“PJSIP/192-0000013e”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“PJSIP/192-0000013e”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“PJSIP/192-0000013e”, “THISDIAL=PJSIP/167”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“PJSIP/192-0000013e”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“PJSIP/192-0000013e”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“PJSIP/192-0000013e”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“PJSIP/192-0000013e”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“PJSIP/192-0000013e”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“PJSIP/192-0000013e”, “THISPART2=PJSIP/167”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“PJSIP/192-0000013e”, “0?Set(THISPART2=DAHDIIP/167)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“PJSIP/192-0000013e”, “NEWDIAL=PJSIP/167&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“PJSIP/192-0000013e”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“PJSIP/192-0000013e”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“PJSIP/192-0000013e”, “THISDIAL=PJSIP/167”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“PJSIP/192-0000013e”, “”) in new stack
– Executing [dstring@macro-dial-one:9] GotoIf(“PJSIP/192-0000013e”, “0?docheck”) in new stack
– Executing [dstring@macro-dial-one:10] NoOp(“PJSIP/192-0000013e”, “Debug: Found PJSIP Destination PJSIP/167”) in new stack
– Executing [dstring@macro-dial-one:11] GotoIf(“PJSIP/192-0000013e”, “0?doset”) in new stack
– Executing [dstring@macro-dial-one:12] NoOp(“PJSIP/192-0000013e”, “Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS”) in new stack
– Executing [dstring@macro-dial-one:13] Set(“PJSIP/192-0000013e”, “THISDIAL=PJSIP/167/sip:167@192.168.4.196:5060”) in new stack
– Executing [dstring@macro-dial-one:14] ExecIf(“PJSIP/192-0000013e”, “0?Set(DIALSTATUS=CHANUNAVAIL)”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“PJSIP/192-0000013e”, “0?skipset”) in new stack
– Executing [dstring@macro-dial-one:16] Set(“PJSIP/192-0000013e”, “DSTRING=PJSIP/167/sip:167@192.168.4.196:5060&”) in new stack
– Executing [dstring@macro-dial-one:17] Set(“PJSIP/192-0000013e”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:18] GotoIf(“PJSIP/192-0000013e”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:19] ExecIf(“PJSIP/192-0000013e”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:20] Set(“PJSIP/192-0000013e”, “DSTRING=PJSIP/167/sip:167@192.168.4.196:5060”) in new stack
– Executing [dstring@macro-dial-one:21] Return(“PJSIP/192-0000013e”, “”) in new stack
– Executing [s@macro-dial-one:29] GotoIf(“PJSIP/192-0000013e”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:30] GotoIf(“PJSIP/192-0000013e”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:31] GosubIf(“PJSIP/192-0000013e”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“PJSIP/192-0000013e”, “DB(CALLTRACE/167)=192”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“PJSIP/192-0000013e”, “”) in new stack
– Executing [s@macro-dial-one:32] Set(“PJSIP/192-0000013e”, “D_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dial-one:33] GosubIf(“PJSIP/192-0000013e”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:34] NoOp(“PJSIP/192-0000013e”, "Blind Transfer: , Attended Transfer: , User: 192, Alert Info: ") in new stack
– Executing [s@macro-dial-one:35] ExecIf(“PJSIP/192-0000013e”, “1?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:36] ExecIf(“PJSIP/192-0000013e”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:37] ExecIf(“PJSIP/192-0000013e”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:38] ExecIf(“PJSIP/192-0000013e”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:39] ExecIf(“PJSIP/192-0000013e”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:40] GosubIf(“PJSIP/192-0000013e”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:41] ExecIf(“PJSIP/192-0000013e”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:42] GosubIf(“PJSIP/192-0000013e”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:43] Set(“PJSIP/192-0000013e”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:44] Set(“PJSIP/192-0000013e”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:45] GotoIf(“PJSIP/192-0000013e”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:46] GotoIf(“PJSIP/192-0000013e”, “0?godial”) in new stack
– Executing [s@macro-dial-one:47] Gosub(“PJSIP/192-0000013e”, “sub-presencestate-display,s,1(167)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“PJSIP/192-0000013e”, “state-available,1”) in new stack
– Goto (sub-presencestate-display,state-available,1)
– Executing [state-available@sub-presencestate-display:1] Set(“PJSIP/192-0000013e”, “PRESENCESTATE_DISPLAY=(Available)”) in new stack
– Executing [state-available@sub-presencestate-display:2] Return(“PJSIP/192-0000013e”, “”) in new stack
– Executing [s@macro-dial-one:48] Set(“PJSIP/192-0000013e”, “CONNECTEDLINE(name,i)=Gate 1(Available)”) in new stack
– Executing [s@macro-dial-one:49] Set(“PJSIP/192-0000013e”, “CONNECTEDLINE(num)=167”) in new stack
– Executing [s@macro-dial-one:50] Set(“PJSIP/192-0000013e”, “D_OPTIONS=HhTtrI”) in new stack
– Executing [s@macro-dial-one:51] Macro(“PJSIP/192-0000013e”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“PJSIP/192-0000013e”, “”) in new stack
– Executing [s@macro-dial-one:52] ExecIf(“PJSIP/192-0000013e”, “0?Set(D_OPTIONS=HhtrII)”) in new stack
– Executing [s@macro-dial-one:53] NoOp(“PJSIP/192-0000013e”, “”) in new stack
– Executing [s@macro-dial-one:54] Dial(“PJSIP/192-0000013e”, “PJSIP/167/sip:167@192.168.4.196:5060,15,HhTtrIb(func-apply-sipheaders^s^1)”) in new stack
– PJSIP/167-0000013f Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“PJSIP/167-0000013f”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“PJSIP/167-0000013f”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“PJSIP/167-0000013f”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:4] ExecIf(“PJSIP/167-0000013f”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:5] While(“PJSIP/167-0000013f”, “0”) in new stack
– Jumping to priority 9
– Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/167-0000013f”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/167-0000013f”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:12] Return(“PJSIP/167-0000013f”, “”) in new stack
== Spawn extension (from-internal, 167, 1) exited non-zero on ‘PJSIP/167-0000013f’
– PJSIP/167-0000013f Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/167/sip:167@192.168.4.196:5060
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Connected line update to PJSIP/192-0000013e prevented.
– PJSIP/167-0000013f is ringing
– PJSIP/167-0000013f is ringing
== Spawn extension (macro-dial-one, s, 54) exited non-zero on ‘PJSIP/192-0000013e’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘PJSIP/192-0000013e’ in macro ‘exten-vm’
== Spawn extension (ext-local, 167, 2) exited non-zero on ‘PJSIP/192-0000013e’
– Executing [h@ext-local:1] Macro(“PJSIP/192-0000013e”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/192-0000013e”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/192-0000013e”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/192-0000013e”, "PJSIP/167-0000013f monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“PJSIP/192-0000013e”, “attendedtransfer-rec-restart.php,PJSIP/167-0000013f,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <PJSIP/192-0000013e>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“PJSIP/192-0000013e”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/192-0000013e’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/192-0000013e’
== Setting global variable ‘SIPDOMAIN’ to ‘192.168.4.1’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
Posts: 1
Participants: 1
@ctorress wrote:
Hi,
I’m running on FreePBX 15.0.16.75 and i would like to have one entry of my IVR to execute the DISA command, but i have multiple DISA entries because each is tied to a different custom context.
The idea is that if the user dials in and uses code 1234 the context “custom-context-1234” will be used but if instead the user uses code 5678 then another context would be used.
I had this setup previously by using a password file where each code was bound to a different context.
While trying to implement this from the UI now i see that the IVR will force me to select the specific DISA code so it i won’t be able to use any of my other DISA configurarations unless i use a different code in the IVR.
I would really like to move away from using custom files and scripts.
Any ideas?
Thanks
Posts: 1
Participants: 1
