@oli170 wrote:

Hi,
actually I test FreePBX distro (14.05.25) together with a Vega 60G analog adapter to get some analog phones running. Works all fine for now with one expection:
I want to achieve to set a call forwarding for a specific extension with a key press on my analog phone attached to the vega (short dial).

Extension to forward: 88
Extension of analog phone: 82
Extension where to forward to: 86

Normally I would do this by dialing *93, followed by the extension to forward, ending with pound, then followed by new destination extension, ending with pound.
But when I dial *93 on my analog phone, no dialog comes up to set my forwarding extension.
On a registered IP-phone Sangoma S405 this works.

But how can I set the call forwarding with one key press? I.e. dialing *93#88#86# without waiting for dialog?
And how do I get the call forwarding running on my analog phone?

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@aspieboy77 wrote:

I’m going to try and make this as easy as possible but need a little help trying to make inbound and outbound phone calls i have attached my log file below please bare with me thank you i’m currently using gotrunk as my trunk provider

[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@func-apply-sipheaders:9] Return(“SIP/gotrunk-00000006”, “”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] app_stack.c: Spawn extension (from-trunk, 2120001234, 1) exited non-zero on ‘SIP/gotrunk-00000006’
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] app_stack.c: SIP/gotrunk-00000006 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] app_dial.c: Called SIP/gotrunk/2120001234
[2019-02-18 07:18:58] NOTICE[2139][C-0000000d] chan_sip.c: Failed to authenticate on INVITE to ‘“Chris Miguez” <sip:Unknown “”"ip address and port number “”>;tag=as3902c43e’
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] app_dial.c: SIP/gotrunk-00000006 is circuit-busy
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-dialout-trunk:32] NoOp(“PJSIP/251-0000000d”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-dialout-trunk:33] GotoIf(“PJSIP/251-0000000d”, “0?continue,1:s-CONGESTION,1”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx_builtins.c: Goto (macro-dialout-trunk,s-CONGESTION,1)
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“PJSIP/251-0000000d”, “RC=21”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“PJSIP/251-0000000d”, “21,1”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx_builtins.c: Goto (macro-dialout-trunk,21,1)
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [21@macro-dialout-trunk:1] Goto(“PJSIP/251-0000000d”, “continue,1”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx_builtins.c: Goto (macro-dialout-trunk,continue,1)
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp(“PJSIP/251-0000000d”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [continue@macro-dialout-trunk:2] ExecIf(“PJSIP/251-0000000d”, “1?Set(CALLERID(number)=251)”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [2120001234@from-internal:8] Macro(“PJSIP/251-0000000d”, “outisbusy,”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-outisbusy:1] Progress(“PJSIP/251-0000000d”, “”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/251-0000000d”, “0?emergency,1”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/251-0000000d”, “0?intracompany,1”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-outisbusy:4] Playback(“PJSIP/251-0000000d”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
[2019-02-18 07:18:58] VERBOSE[9088][C-0000000d] file.c: <PJSIP/251-0000000d> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] file.c: <PJSIP/251-0000000d> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/251-0000000d”, “hangupcall”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/251-0000000d”, “1?theend”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/251-0000000d”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/251-0000000d”, " monior file= ") in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-hangupcall:5] AGI(“PJSIP/251-0000000d”, “attendedtransfer-rec-restart.php,”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] res_agi.c: <PJSIP/251-0000000d>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“PJSIP/251-0000000d”, “”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/251-0000000d’ in macro ‘hangupcall’
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/251-0000000d’
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] app_stack.c: PJSIP/251-0000000d Internal Gosub(crm-hangup,s,1) start
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/251-0000000d”, “Sending Hangup to CRM”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/251-0000000d”, “HANGUP CAUSE: 21”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/251-0000000d”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/251-0000000d”, “MASTER CHANNEL: 1550474338.19 = 1550474338.19”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/251-0000000d”, “0?return”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/251-0000000d”, “__CRM_HANGUP=1”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/251-0000000d”, “sangomacrm.agi”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] res_agi.c: <PJSIP/251-0000000d>AGI Script sangomacrm.agi completed, returning 0
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/251-0000000d”, “”) in new stack
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/251-0000000d’
[2019-02-18 07:18:59] VERBOSE[9088][C-0000000d] app_stack.c: PJSIP/251-0000000d Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

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@johnk wrote:

fresh install 2/17 of sng7 64bit 1805-2. I enter email and password and get response noted in title. there are no other options that appear. I enter my password in a box requesting same. Then a box pops up requesting password. I enter this and press login button (only other button is abort) and get the error.

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@brunozerves wrote:

Hello friends, I am new here in the Forum, I am installing a new Freepbx for the first time, with a 6 line SIP trunk… What I would need to do and I would like to know if it is possible is this:

I’m going to advertise 3 of those numbers to get calls, but when someone calls, I need to forward that call to a customer’s number and record that call …

It’s possible? Can you help me?

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@Chimnysweep wrote:

Hi,

Old Aastra phone died. Client would now like to get a cordless phone.

Anyone have any experience with the Yealink W60B DECT IP Base Station and W56H handset?

Thanks,
Westley

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@tippetts wrote:

Good day

Recently ran module updates. In the past they usually run with no issues.Here is the error I receive. Right now I am unable to apply config and endpoint manager is non functional. Assistance is appriciated.

exit: 1
Exception: Unable to locate the FreePBX BMO Class 'Core’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install core 2) fwconsole ma enable core in file /var/www/html/admin/libraries/BMO/Self_Helper.class.php on line 216
Stack trace:

1. Exception->() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:216
2. FreePBX\Self_Helper->loadObject() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:104
3. FreePBX\Self_Helper->autoLoad() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:37
4. FreePBX\Self_Helper->__get() /var/www/html/admin/modules/certman/Certman.class.php:39
5. FreePBX\modules\Certman->__construct() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:125
6. FreePBX\Self_Helper->autoLoad() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:37
7. FreePBX\Self_Helper->__get() /var/www/html/admin/libraries/BMO/Hooks.class.php:294
8. FreePBX\Hooks->preloadBMOModules() /var/www/html/admin/libraries/BMO/Hooks.class.php:39
9. FreePBX\Hooks->updateBMOHooks() /var/lib/asterisk/bin/retrieve_conf:61

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@cedric22 wrote:

Greetings everyone,

Long time lurker/reader 1st time poster.

I’m in need of your input/advice on my current setup:

1. Have a main number that hits one Call Flow which is our Switch to Night Mode
2. Normal Flow for Night Mode = Queue, Override Flow = Announcement (Nigh Mode after hours)
3. Only have one Queue, this queue has many agents.
4. Queue is set to fail back on itself indefinitely until calls are answered.

So far so good, however here are a few key requirements.
When calls come through but not answered go back to Queue
When calls come through to Agent directly but busy go back to Queue
Each Agent has a direct number, so having call forwards helps in the event that they are away or busy.

I need Team leaders to have access to UCP to control call forward of their staff.
Pointing the call forwards to the Queue works well when Call Flow is not active.

Obviously when Call Flow is active, I have issues if people call the agents directly,
As after the call forward for no answer kicks in and points to Queue it will ring the Queue and make all the phones ring.
However for the life of me I can’t figure out a way to call forward to the Call Flow via UCP.

Current setup:
All Agents have their Call Forward Busy set to Queue in UCP
All Agents have their No Answer set to Call Flow in their Advance tab of their Extension under Optional Destinations.

This somewhat works but it takes away the control from my Team Leaders.

Maybe there is a better way for me to set this up both Queue wise & Night mode wise.
I have noticed that doing a No Answer or Busy to Call Flow in the Advance Tab vs to Queue, as an external caller the ringing changes when it does the switch.

I’ve had a few outsiders point that out to me.

Thank you for giving me your thoughts on this.
Definitely willing to learn.

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@nsumner wrote:

Okay, this is clearly configuration related but.

I have a softphone for several people. Outside of the office it works great. Inside the office it works great. Connect to our VPN (OpenVPN) it doesn’t work.

Pings work, but obviously, I’m hitting some sort of NAT issue but no matter when I try I don’t seem to be able to solve it. Does anybody have any ideas or anything else? I don’t mind opening a support ticket and paying but I’m not sure if they can help me with this.

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@GSnover wrote:

I have a Valcom paging system set up behind a Sangoma 7/FreePBX 14 instance using a Digium AEX-410 with a single FXO port that is working fine - but I want to Virtualize.

So I set up a GrandStream HT503 with an FXO port to take over for the Digium Card.

The weird thing is the Valcom requires DTMF to select the zone to be paged before the page will go out - this was easily done with the Digium card - in this case the All-Page code is #10 - I have had that working for years with the Digium, but when I try to do it with the Grandstream, it fails.

It’s a School and they use it for the Class Bells with a CRON Job and a Call-File that pages and plays a beep.

Where I am stuck is if I set up the FXO port as an extension, it won’t pass (and dial) the #10 - but if you call the extension and press #10 yourself, it works fine - so Paging is working fine, but the Class-Bell system is failing.

Has anyone else tried anything similar and gotten it to work? I think it should be possible, but I just can’t get the coding correct.

If I set up the FXO as a Trunk, I get an Auth Reject when I try to pass the #10 in the call file - Here is the Call-File that is working for the Digium Card:

Channel: DAHDI/1/#10
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: custom-class-bell
Extension: 10

So my assumption was to just change it to the Extension that I had created:

Channel: SIP/2323/#10
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: custom-class-bell
Extension: 2323

But like I said, it doesn’t like the #10 as the Dial String.

Any ideas?

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@oleggram wrote:

What is the right configuration of SIP trunk between FreePBX and Yeastar U-100?
“Show SIP registry” shows my trunk “not authenticated”. I tried a lot of combinations with and w/o password mentioned on the forums but w/o result. If i tried U-100 w/o password it shows Registered during approx. 20sec and then “Request sent”. PjSIP trunk works only from FreePBX to U-100 but not vice versa.
My simple config for sip is:
host=192.168.1.201
username=Asterisk
secret=asterisk
type=friend
context=from-internal
qualify=yes

Incoming register string:
Asterisk:asterisk@192.168.1.201

Similar configuration works with Digium gateway w/o problem

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@Hawkeye wrote:

When running asterisk-version-switch and choosing 3 for Asterisk 16, asterisk16-core-16.0.0-1.sng7.x86_64 failed to install causing the switch to asterisk16 unusable.

PBX Version: 12.7.5-1902-3.sng7

**# asterisk-version-switch**

Pick the Asterisk Version you would like to change to.
Press 1 and the Enter key for Asterisk 13 (LTS) (With Opus and G729 codecs)
Press 2 and the Enter key for Asterisk 15 (With Opus and G729 codecs)
Press 3 and the Enter key for Asterisk 16 (LTS) (With Opus and G729 codecs)
Press 9 and the Enter key to exit and not change your Asterisk Version
3

You picked asterisk16

Clearing yum cache
Loaded plugins: fastestmirror, versionlock
Cleaning repos: sng-base sng-epel sng-extras sng-pkgs sng-updates
Cleaning up everything
Maybe you want: rm -rf /var/cache/yum, to also free up space taken by orphaned data from disabled or removed repos
Cleaning up list of fastest mirrors

Switching to Asterisk 16

Loaded plugins: fastestmirror, versionlock

Determining fastest mirrors
sng-base | 3.6 kB 00:00:00
sng-epel | 3.2 kB 00:00:00
sng-extras | 3.4 kB 00:00:00
sng-pkgs | 2.9 kB 00:00:00
sng-updates | 3.4 kB 00:00:00
(1/8): sng-epel/7/x86_64/group_gz | 88 kB 00:00:00
(2/8): sng-base/7/x86_64/group_gz | 166 kB 00:00:00
(3/8): sng-pkgs/7/x86_64/primary_db | 610 kB 00:00:00
(4/8): sng-epel/7/x86_64/updateinfo | 933 kB 00:00:00
(5/8): sng-base/7/x86_64/primary_db | 5.9 MB 00:00:00
(6/8): sng-extras/7/x86_64/primary_db | 173 kB 00:00:00
(7/8): sng-epel/7/x86_64/primary | 3.6 MB 00:00:00
(8/8): sng-updates/7/x86_64/primary_db | 4.3 MB 00:00:00
sng-epel 12642/12642

Package jansson-2.10-1.el7.x86_64 already installed and latest version
Package jansson-devel-2.10-1.el7.x86_64 already installed and latest version
Package jansson-devel-doc-2.10-1.el7.noarch already installed and latest version
Package sangoma-pbx-1902-3.sng7.noarch already installed and latest version
–> Running transaction check
—> Package asterisk13.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-addons.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-addons-bluetooth.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-addons-core.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-addons-mysql.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-addons-ooh323.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-curl.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-dahdi.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-doc.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-flite.x86_64 0:2.4-4_2505af1.sng7 will be erased
—> Package asterisk13-g729.x86_64 0:1711-2.sng7 will be erased
—> Package asterisk13-odbc.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-ogg.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-resample.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk13-voicemail.x86_64 0:13.22.0-1.sng7 will be erased
—> Package asterisk16.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-addons.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-addons-bluetooth.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-addons-core.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-addons-mysql.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-addons-ooh323.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-core.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-curl.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-dahdi.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-doc.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-flite.x86_64 0:2.4-2_2505af1.sng7 will be installed
—> Package asterisk16-g729.x86_64 0:1711-2.sng7 will be installed
—> Package asterisk16-odbc.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-ogg.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-resample.x86_64 0:16.0.0-1.sng7 will be installed
—> Package asterisk16-voicemail.x86_64 0:16.0.0-1.sng7 will be installed
–> Finished Dependency Resolution
Success resolving dependencies
–> Running transaction check
–> Finished Dependency Resolution

=======================================================================================================================================
Package Arch Version Repository Size

Installing:
asterisk16 x86_64 16.0.0-1.sng7 sng-pkgs 2.8 k
asterisk16-addons x86_64 16.0.0-1.sng7 sng-pkgs 1.9 k
asterisk16-addons-bluetooth x86_64 16.0.0-1.sng7 sng-pkgs 36 k
asterisk16-addons-core x86_64 16.0.0-1.sng7 sng-pkgs 25 k
asterisk16-addons-mysql x86_64 16.0.0-1.sng7 sng-pkgs 37 k
asterisk16-addons-ooh323 x86_64 16.0.0-1.sng7 sng-pkgs 340 k
asterisk16-core x86_64 16.0.0-1.sng7 sng-pkgs 5.1 M
asterisk16-curl x86_64 16.0.0-1.sng7 sng-pkgs 20 k
asterisk16-dahdi x86_64 16.0.0-1.sng7 sng-pkgs 284 k
asterisk16-doc x86_64 16.0.0-1.sng7 sng-pkgs 12 k
asterisk16-flite x86_64 2.4-2_2505af1.sng7 sng-pkgs 8.2 k
asterisk16-g729 x86_64 1711-2.sng7 sng-pkgs 181 k
asterisk16-odbc x86_64 16.0.0-1.sng7 sng-pkgs 62 k
asterisk16-ogg x86_64 16.0.0-1.sng7 sng-pkgs 8.1 k
asterisk16-resample x86_64 16.0.0-1.sng7 sng-pkgs 13 k
asterisk16-voicemail x86_64 16.0.0-1.sng7 sng-pkgs 80 k
Removing:
asterisk13 x86_64 13.22.0-1.sng7 @sng-pkgs 0.0
asterisk13-addons x86_64 13.22.0-1.sng7 @sng-pkgs 0.0
asterisk13-addons-bluetooth x86_64 13.22.0-1.sng7 @sng-pkgs 107 k
asterisk13-addons-core x86_64 13.22.0-1.sng7 @sng-pkgs 71 k
asterisk13-addons-mysql x86_64 13.22.0-1.sng7 @sng-pkgs 113 k
asterisk13-addons-ooh323 x86_64 13.22.0-1.sng7 @sng-pkgs 1.9 M
asterisk13-curl x86_64 13.22.0-1.sng7 @sng-pkgs 64 k
asterisk13-dahdi x86_64 13.22.0-1.sng7 @sng-pkgs 969 k
asterisk13-doc x86_64 13.22.0-1.sng7 @sng-pkgs 8.6 k
asterisk13-flite x86_64 2.4-4_2505af1.sng7 @sng-pkgs 15 k
asterisk13-g729 x86_64 1711-2.sng7 @sng-pkgs 597 k
asterisk13-odbc x86_64 13.22.0-1.sng7 @sng-pkgs 213 k
asterisk13-ogg x86_64 13.22.0-1.sng7 @sng-pkgs 16 k
asterisk13-resample x86_64 13.22.0-1.sng7 @sng-pkgs 26 k
asterisk13-voicemail x86_64 13.22.0-1.sng7 @sng-pkgs 246 k

Transaction Summary

Install 16 Packages
Remove 15 Packages

Total download size: 6.2 M
Downloading packages:
(1/16): asterisk16-addons-16.0.0-1.sng7.x86_64.rpm | 1.9 kB 00:00:00
(2/16): asterisk16-16.0.0-1.sng7.x86_64.rpm | 2.8 kB 00:00:00
(3/16): asterisk16-addons-bluetooth-16.0.0-1.sng7.x86_64.rpm | 36 kB 00:00:00
(4/16): asterisk16-addons-mysql-16.0.0-1.sng7.x86_64.rpm | 37 kB 00:00:00
(5/16): asterisk16-addons-core-16.0.0-1.sng7.x86_64.rpm | 25 kB 00:00:00
(6/16): asterisk16-addons-ooh323-16.0.0-1.sng7.x86_64.rpm | 340 kB 00:00:00
(7/16): asterisk16-curl-16.0.0-1.sng7.x86_64.rpm | 20 kB 00:00:00
(8/16): asterisk16-dahdi-16.0.0-1.sng7.x86_64.rpm | 284 kB 00:00:00
(9/16): asterisk16-doc-16.0.0-1.sng7.x86_64.rpm | 12 kB 00:00:00
(10/16): asterisk16-core-16.0.0-1.sng7.x86_64.rpm | 5.1 MB 00:00:00
(11/16): asterisk16-flite-2.4-2_2505af1.sng7.x86_64.rpm | 8.2 kB 00:00:00
(12/16): asterisk16-odbc-16.0.0-1.sng7.x86_64.rpm | 62 kB 00:00:00
(13/16): asterisk16-g729-1711-2.sng7.x86_64.rpm | 181 kB 00:00:00
(14/16): asterisk16-resample-16.0.0-1.sng7.x86_64.rpm | 13 kB 00:00:00
(15/16): asterisk16-voicemail-16.0.0-1.sng7.x86_64.rpm | 80 kB 00:00:00
(16/16): asterisk16-ogg-16.0.0-1.sng7.x86_64.rpm | 8.1 kB 00:00:00

Total 14 MB/s | 6.2 MB 00:00:00
Running transaction check
Running transaction test
Transaction test succeeded
Running transaction

Installing : asterisk16-core-16.0.0-1.sng7.x86_64 1/31
Error unpacking rpm package asterisk16-core-16.0.0-1.sng7.x86_64
error: unpacking of archive failed on file /etc/logrotate.d/asterisk;5c6c321f: cpio: rename
Installing : asterisk16-addons-core-16.0.0-1.sng7.x86_64 2/31
error: asterisk16-core-16.0.0-1.sng7.x86_64: install failed
Installing : asterisk16-addons-mysql-16.0.0-1.sng7.x86_64 3/31
Installing : asterisk16-dahdi-16.0.0-1.sng7.x86_64 4/31
Installing : asterisk16-addons-ooh323-16.0.0-1.sng7.x86_64 5/31
Installing : asterisk16-doc-16.0.0-1.sng7.x86_64 6/31
Installing : asterisk16-voicemail-16.0.0-1.sng7.x86_64 7/31
Installing : asterisk16-16.0.0-1.sng7.x86_64 8/31
Installing : asterisk16-addons-bluetooth-16.0.0-1.sng7.x86_64 9/31
Installing : asterisk16-addons-16.0.0-1.sng7.x86_64 10/31
Installing : asterisk16-flite-2.4-2_2505af1.sng7.x86_64 11/31
Installing : asterisk16-resample-16.0.0-1.sng7.x86_64 12/31
Installing : asterisk16-odbc-16.0.0-1.sng7.x86_64 13/31
Installing : asterisk16-curl-16.0.0-1.sng7.x86_64 14/31
Installing : asterisk16-g729-1711-2.sng7.x86_64 15/31
Installing : asterisk16-ogg-16.0.0-1.sng7.x86_64 16/31
Erasing : asterisk13-addons-13.22.0-1.sng7.x86_64 17/31
Erasing : asterisk13-addons-mysql-13.22.0-1.sng7.x86_64 18/31
Erasing : asterisk13-flite-2.4-4_2505af1.sng7.x86_64 19/31
Erasing : asterisk13-13.22.0-1.sng7.x86_64 20/31
Erasing : asterisk13-dahdi-13.22.0-1.sng7.x86_64 21/31
Erasing : asterisk13-doc-13.22.0-1.sng7.x86_64 22/31
Erasing : asterisk13-g729-1711-2.sng7.x86_64 23/31
Erasing : asterisk13-voicemail-13.22.0-1.sng7.x86_64 24/31
Erasing : asterisk13-addons-core-13.22.0-1.sng7.x86_64 25/31
Erasing : asterisk13-addons-bluetooth-13.22.0-1.sng7.x86_64 26/31
Erasing : asterisk13-addons-ooh323-13.22.0-1.sng7.x86_64 27/31
Erasing : asterisk13-ogg-13.22.0-1.sng7.x86_64 28/31
Erasing : asterisk13-curl-13.22.0-1.sng7.x86_64 29/31
Erasing : asterisk13-odbc-13.22.0-1.sng7.x86_64 30/31
Erasing : asterisk13-resample-13.22.0-1.sng7.x86_64 31/31
Verifying : asterisk16-flite-2.4-2_2505af1.sng7.x86_64 1/31
Verifying : asterisk16-addons-core-16.0.0-1.sng7.x86_64 2/31
Verifying : asterisk16-dahdi-16.0.0-1.sng7.x86_64 3/31
Verifying : asterisk16-addons-mysql-16.0.0-1.sng7.x86_64 4/31
Verifying : asterisk16-16.0.0-1.sng7.x86_64 5/31
Verifying : asterisk16-resample-16.0.0-1.sng7.x86_64 6/31
Verifying : asterisk16-odbc-16.0.0-1.sng7.x86_64 7/31
Verifying : asterisk16-curl-16.0.0-1.sng7.x86_64 8/31
Verifying : asterisk16-addons-ooh323-16.0.0-1.sng7.x86_64 9/31
Verifying : asterisk16-doc-16.0.0-1.sng7.x86_64 10/31
Verifying : asterisk16-voicemail-16.0.0-1.sng7.x86_64 11/31
Verifying : asterisk16-addons-bluetooth-16.0.0-1.sng7.x86_64 12/31
Verifying : asterisk16-addons-16.0.0-1.sng7.x86_64 13/31
Verifying : asterisk16-g729-1711-2.sng7.x86_64 14/31
Verifying : asterisk16-ogg-16.0.0-1.sng7.x86_64 15/31
Verifying : asterisk13-resample-13.22.0-1.sng7.x86_64 16/31
Verifying : asterisk13-odbc-13.22.0-1.sng7.x86_64 17/31
Verifying : asterisk13-curl-13.22.0-1.sng7.x86_64 18/31
Verifying : asterisk13-addons-13.22.0-1.sng7.x86_64 19/31
Verifying : asterisk13-addons-bluetooth-13.22.0-1.sng7.x86_64 20/31
Verifying : asterisk16-core-16.0.0-1.sng7.x86_64 21/31
Verifying : asterisk13-addons-mysql-13.22.0-1.sng7.x86_64 22/31
Verifying : asterisk13-ogg-13.22.0-1.sng7.x86_64 23/31
Verifying : asterisk13-flite-2.4-4_2505af1.sng7.x86_64 24/31
Verifying : asterisk13-addons-core-13.22.0-1.sng7.x86_64 25/31
Verifying : asterisk13-voicemail-13.22.0-1.sng7.x86_64 26/31
Verifying : asterisk13-doc-13.22.0-1.sng7.x86_64 27/31
Verifying : asterisk13-addons-ooh323-13.22.0-1.sng7.x86_64 28/31
Verifying : asterisk13-g729-1711-2.sng7.x86_64 29/31
Verifying : asterisk13-dahdi-13.22.0-1.sng7.x86_64 30/31
Verifying : asterisk13-13.22.0-1.sng7.x86_64 31/31

Removed:
asterisk13.x86_64 0:13.22.0-1.sng7 asterisk13-addons.x86_64 0:13.22.0-1.sng7
asterisk13-addons-bluetooth.x86_64 0:13.22.0-1.sng7 asterisk13-addons-core.x86_64 0:13.22.0-1.sng7
asterisk13-addons-mysql.x86_64 0:13.22.0-1.sng7 asterisk13-addons-ooh323.x86_64 0:13.22.0-1.sng7
asterisk13-curl.x86_64 0:13.22.0-1.sng7 asterisk13-dahdi.x86_64 0:13.22.0-1.sng7
asterisk13-doc.x86_64 0:13.22.0-1.sng7 asterisk13-flite.x86_64 0:2.4-4_2505af1.sng7
asterisk13-g729.x86_64 0:1711-2.sng7 asterisk13-odbc.x86_64 0:13.22.0-1.sng7
asterisk13-ogg.x86_64 0:13.22.0-1.sng7 asterisk13-resample.x86_64 0:13.22.0-1.sng7
asterisk13-voicemail.x86_64 0:13.22.0-1.sng7

Installed:
asterisk16.x86_64 0:16.0.0-1.sng7 asterisk16-addons.x86_64 0:16.0.0-1.sng7
asterisk16-addons-bluetooth.x86_64 0:16.0.0-1.sng7 asterisk16-addons-core.x86_64 0:16.0.0-1.sng7
asterisk16-addons-mysql.x86_64 0:16.0.0-1.sng7 asterisk16-addons-ooh323.x86_64 0:16.0.0-1.sng7
asterisk16-curl.x86_64 0:16.0.0-1.sng7 asterisk16-dahdi.x86_64 0:16.0.0-1.sng7
asterisk16-doc.x86_64 0:16.0.0-1.sng7 asterisk16-flite.x86_64 0:2.4-2_2505af1.sng7
asterisk16-g729.x86_64 0:1711-2.sng7 asterisk16-odbc.x86_64 0:16.0.0-1.sng7
asterisk16-ogg.x86_64 0:16.0.0-1.sng7 asterisk16-resample.x86_64 0:16.0.0-1.sng7
asterisk16-voicemail.x86_64 0:16.0.0-1.sng7

Failed:
asterisk16-core.x86_64 0:16.0.0-1.sng7

There were non-fatal errors in the transaction
Finished Transaction

Leaving Shell

asterisk16 has now been verified to be installed

Restarting Asterisk…
Running FreePBX shutdown…

Stopping UCP Node Server
[>---------------------------] < 1 sec
Stopped UCP Node Server
Stopping Chat Server
Stopped Chat Server
Zulu Server is not running
Shutting down Asterisk Gracefully. Will forcefully kill after 30 seconds.
Press C to Cancel
Press N to shut down NOW
[============================] < 1 sec
Running FreePBX startup…
Running Asterisk pre from Sysadmin module
Running Sysadmin Hooks
Restarting fail2ban
fail2ban Restarted
Updating License Information for 17562579
Checking Vpn server
Starting Asterisk…
[============================] 3 secs
Asterisk Started
Running Asterisk post from Ucp module
Starting UCP Node Server…
[>---------------------------] < 1 sec
Started UCP Node Server. PID is 6872
Running Asterisk post from Xmpp module
Resetting PBX Users Failed: The command “node /var/www/html/admin/modules/xmpp/node/resetpbxusers.js” failed.

Exit Code: 1(General error)

Working directory: /root

Output:

Error Output:

Unable to connect to asterisk!

/var/www/html/admin/modules/xmpp/node/node_modules/freepbx/lib/freepbx.js:103
throw “There was an error with Asterisk Manager Connection, is Asterisk running?”;
^
There was an error with Asterisk Manager Connection, is Asterisk running?

Running Asterisk post from Zulu module
Asterisk is not connected
Regenerate FreePBX Dialplan…
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1

Chowning all needed files
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions…
Setting base permissions…chmod: changing permissions of â/var/www/html/admin/views/footer_content.phpâ: Operation not permitted
chown: changing ownership of â/var/www/html/admin/views/footer_content.phpâ: Operation not permitted
Done
Setting specific permissions…
64706 [============================]
Finished setting permissions

Any ideas as to why asterisk16-core-16.0.0-1.sng7.x86_64 is failing?
Thanks.

Posts: 1

Participants: 1

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@Brandonwinstead wrote:

I am upgrading a freepbx distro from 13 to 14 using the upgrade tool. After the system reboots the second time I have video for awhile then the video just stops and never comes back on. After the upgrade is complete the machine functions with no video output. I can access the machine VIA SSH and the web GUI. I’ve read online I can set the video in grub to nomodeset, and that should fix the issue. Where do I put the nomodeset in grub? Can I edit the grub fia ssh into the machine or can I only get it on reboot?

I’m running this as a VM on proxmox .

Posts: 1

Participants: 1

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@kwriley87 wrote:

Hello-

We are using FreePBX Distro 14 with Asterisk 13.

When a user dials *32 to blacklist the last caller, it says the last caller was a caller ID number that was not the actual last caller.

We have updated all modules, and performed OS updates but the behavior still occurs.

Can anyone help?

Posts: 1

Participants: 1

Read full topic

@comtech wrote:

I have nine numbers:
111.111.1111
222.222.2222
…
999.999.9999

I want to store a variable for use, by number. I was thinking AstDB would be the right tool for the job, but wants quite sure how to do it.

The variable will come from a .txt file in in the /etc/asterisk folder.
For example:
the variable for 111.111.1111 would come from the /etc/asterisk/1111111111.txt file.

The .txt file would only contain the variable, the entire content of the file is the variable.
For example, 1111111111.txt would contain:
This whole sentence is the variable.

Questions:
What’s a good way to structure this from a family key perspective? I am having a hard time with this.

Given the command, what’s the DB update command I should use to set the data?

Thanks in advance for any ideas you may have.

Posts: 1

Participants: 1

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@mtrudel34 wrote:

Hey there, alright so I’ve been working on this project of mine in my spare time for the past week or two and I have finally been able to get asterisk up and running (woohoo!). All of my inbound and outbound calls are working as expected. I have been using freePBX to set everything up and I have been looking forward to taking advantage of the callback feature there-within. I was able to get the callback feature up and running but I am a little disappointed with how the system works and I would like to customize it just a tad.

There are two main issues I am having that I would like to change:

• The first is that rather than terminating the call, the callback feature accepts the call and
  gives the outside caller a busy signal and the caller must then manually end the call. This
  results in the callback sometimes being missed because the line remains tied up on the
  outside end for too long. I would instead like to just have the call terminated and hung up
  immediately to fix this issue.

• The second thing I would like to modify is that when the callback goes through, the outside
  caller accepts the callback and the call is then forwarded to the extension designated, the
  extension rings as it should but the caller ID shows up as UNKNOWN. I would instead prefer
  to have the caller ID read the number of the outside caller so that the party at the extension
  can see who is calling before accepting.

From what I can tell this can be modified to work as I would like with some custom coding or perhaps there is another way? I am not well versed in the Asterisk server files and what will need to be changed. I will continue researching until I find a solution that works but this will take sometime. In the mean time I figured it wouldn’t hurt to put up a quick post on here to help speed me along the road to success. Any help is appreciated and thank in advance!

Posts: 2

Participants: 2

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@aspieboy77 wrote:

hello my name is chris and i’m running FREEPBX VERSION 14.0.5.25 and cannot get the outbound caller id working they work in bound when i receive a phone call but when i call out from my phone it says unknown number. on multiple different phones how can i fix this. please can you explain in small steps im still learning freepbx but have no issues with using a computer or setting them up i repair computers for a living as a side job just not good at this phone stuff thanks in advance

Posts: 1

Participants: 1

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@aspieboy77 wrote:

Hello i have a few questions related to freepbx i’m looking at starting a company that offers phone services to company’s and small business for around 20 to 29 us dollars its my hope that i can use freepbx to accomplish this my question is i need some advice on trunk providers that i can get bulk numbers from for freepbx to use in that use case and also just if i’m going in the right direction. if someone could jump on the bandwagon and help me with these questions that would be willing to hold my hand and explain it to me that would be grate thanks

Posts: 5

Participants: 3

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@jerryriggin wrote:

Assume all FreePBX 14. I have a queue on FreePBX-01. I have extensions registered on FreePBX-02. I have been trying to work out how to have extensions registered to FreePBX-02 as queue members in the FreePBX-01 queue. I was looking for a way to login “1001@FreePBX-02” without creating a trunk or custom extensions on FreePBX-01.

Is there a way to do this?

Posts: 1

Participants: 1

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@oldtimer1955 wrote:

I am building a new installation of FreePBX (FreePBX 14.0.5.25) on RasPBX
All the modules I am using are up to date.
The system is fully configured and was tested working fine and the Dashboard was fully visible.
I powered down, moved the physical location of the box, reconnected to the network and powered up.
Everything seems to be working fine, I can see the modules and configuration and calls are being handled but the problem is the DASHBOARD IS BLANK.

I get a single line of text:
0 System Admin 14.0.31 Copyright 2019 by Sangoma Technologies Inc., A… (and the legal guff).
Clearly a page is being served by FreePBX but there is nothing else - no navigation or context menus.

I was on Dashboard 14.0.4.4
I have tried rolling back to 14.0.4.0, but no difference.
I have tried disabling the SIPstation module (as suggested by someone), but no difference.
I have tried rebooting Image may be NSFW.
Clik here to view.

http://192.168.2.171/admin/config.php has that single line of text, as does:
http://192.168.2.171/admin/config.php?display=index

If I go to say, http://192.168.2.171/admin/config.php?display=trunks
then the display is fine and I can access all modules, settings etc from the GUI

Can anyone suggest what to do to get the Dashboard to display?

Thanks.

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Participants: 1

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@moehammond wrote:

I can make all kinds of calls, inbound from outside to DID assigned on Skype for Business through FreePBX, I can call from FreePBX extension to Skype for Business ext, I can call from that extension to any number outside.

But only when I am trying to call from Skype for Business to outside through freePBX it says the number doesn’t exist check the number and try again.

I am calling from INVITE 3072481662 to +13201009553
The 307 number is assigned on Skype for Business DID.
Target Number: 3201009553 is PSTN Number

My Skype for Business server is 192.168.20.25
My FreePBX IP is 10.0.1.3

Call Logs are attached:>

freepbx*CLI> sip set debug on
SIP Debugging enabled

<— SIP read from TCP:192.168.20.25:54060 —>
INVITE sip:+13201009553@10.0.1.3;user=phone SIP/2.0
FROM: "adminM"sip:+13072481662;ext=44124@SkypeForBusiness.net;user=phone;epid=075CFA3A59;tag=ba65c54576
TO: sip:+13201009553@10.0.1.3;user=phone
CSEQ: 83955 INVITE
CALL-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bKdcd9998b
CONTACT: sip:SBG-FE01.SkypeForBusiness.net:5060;transport=Tcp;maddr=192.168.20.25;ms-opaque=72f0d0298b3a73fe
CONTENT-LENGTH: 345
SUPPORTED: 100rel
USER-AGENT: RTCC/6.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
P-ASSERTED-IDENTITY: "adminM"sip:adminm@SkypeForBusiness.net,tel:+13072481662;ext=44124
Privacy: id
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE

v=0
o=- 27555 1 IN IP4 192.168.20.25
s=session
c=IN IP4 192.168.20.25
b=CT:1000
t=0 0
m=audio 54564 RTP/AVP 97 101 13 0 8
c=IN IP4 192.168.20.25
a=rtcp:54565
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
— (16 headers 18 lines) —
Sending to 192.168.20.25:54060 (NAT)
Sending to 192.168.20.25:54060 (NAT)
Using INVITE request as basis request - b2e83a43-1878-4d99-9d59-c86ade39cc93
No matching peer for ‘+13072481662;ext=44124’ from ‘192.168.20.25:54060’
Found RTP audio format 97
Found RTP audio format 101
Found RTP audio format 13
Found RTP audio format 0
Found RTP audio format 8
Found unknown media description format RED for ID 97
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 13
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.20.25:54564
Looking for +13201009553 in from-sip-external (domain 10.0.1.3)
sip_route_dump: route/path hop: sip:SBG-FE01.SkypeForBusiness.net:5060;transport=Tcp;maddr=192.168.20.25;ms-opaque=72f0d0298b3a73fe

<— Transmitting (NAT) to 192.168.20.25:54060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bKdcd9998b;received=192.168.20.25;rport=54060
From: "adminM"sip:+13072481662;ext=44124@SkypeForBusiness.net;user=phone;epid=075CFA3A59;tag=ba65c54576
To: sip:+13201009553@10.0.1.3;user=phone
Call-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
CSeq: 83955 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:+13201009553@66.23.202.104:5060;transport=tcp
Content-Length: 0

<------------>
[2019-02-20 14:05:18] WARNING[18850][C-000001c8]: Ext. s:6 @ from-sip-external: “Rejecting unknown SIP connection from 192.168.20.25”
Audio is at 19398
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.20.25:54060 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bKdcd9998b;received=192.168.20.25;rport=54060
From: "adminM"sip:+13072481662;ext=44124@SkypeForBusiness.net;user=phone;epid=075CFA3A59;tag=ba65c54576
To: sip:+13201009553@10.0.1.3;user=phone;tag=as3fb52f4e
Call-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
CSeq: 83955 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:+13201009553@66.23.202.104:5060;transport=tcp
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1366665202 1366665202 IN IP4 66.23.202.104
s=Asterisk PBX 13.22.0
c=IN IP4 66.23.202.104
t=0 0
m=audio 19398 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<— SIP read from TCP:192.168.20.25:54060 —>
ACK sip:+13201009553@66.23.202.104:5060;transport=tcp SIP/2.0
FROM: sip:+13072481662;ext=44124@SkypeForBusiness.net;user=phone;epid=075CFA3A59;tag=ba65c54576
TO: sip:+13201009553@10.0.1.3;user=phone;tag=as3fb52f4e
CSEQ: 83955 ACK
CALL-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bK69fead56
CONTENT-LENGTH: 0
USER-AGENT: RTCC/6.0.0.0 MediationServer

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