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Very slow reload on extension submit

February 11, 2019, 10:20 am
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@CReynolds wrote:

Hello!

One of my systems has recently developed a very slow reload when I click submit on building a new PJSIP extension. It now takes almost 5 minutes for the page to reload after clicking submit. I have not noticed anything else acting slow. Only on building extensions. The system is normal otherwise.

This system is running FreePBX 13.0.195.26
It is a VM with 2 CPU cores and 8GB RAM (Load average: 0.33 0.25 0.20 / 1.5GB RAM usage)
I have 447 extensions built currently but only about 40 are actually registered yet.
The system is about 1.5 years old.
All FreePBX modules are up to date.
I rebooted the system this morning and the problem persists.

I am out of ideas to look at for this so any help or suggestions would be appreciated.

Thanks!

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No Interfaces Found - Fresh FreePBX 14 Distro

February 11, 2019, 12:09 pm
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@mvogel4949 wrote:

On two separate systems after a fresh V14 install I get a No Interfaces Found message at the command line when I log in. I check etc/sysconfig/network-scripts folder and there is no ifcfg-eth0 made so I make it and even with eth0 info there the system will not boot with an IP. The LAN port is active in the BIOS. Thoughts?

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Concurrent Calls Module 14.0.5

February 11, 2019, 12:29 pm
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@TurboCow wrote:

I’m having problems getting this module to work. Everytime I click on it, it goes to this error screen that says:

Whoops \ Exception \ ErrorException (E_DEPRECATED)

mysql_connect(): The mysql extension is deprecated and will be removed in the future: use mysqli or PDO instead

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Duplication in Asteriskcdrdb Records

February 11, 2019, 2:47 pm
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@faisalkhan wrote:

Hi,
I have enabled odbc connector for database populated for the CDRs but I am figuring out some strange thing.

it’s placing two entries for the same call in the database.

Actually each call has duplicate entry now.

I need help in this.

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Call Recording by IVR

February 11, 2019, 6:24 pm
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@cnissarte wrote:

Hello to the community,

i am pretty new in the freepbx world. My company presently used FreePBX for the IVR features, all the phone are managed by an other pbx. Inbound call flow from the pstn to the ivr, and after the selection, to a pbx phone by the trunk.

we received a request to record call for specific ivr, not all the ivr, 2 exactly. for this configuration, presently the RTP flow by Freepbx when an inbound call is received and managed by the IVR.

Do you think that it can be possible to configure it in this way? if yes, any advice to share?

Thanks a lot for any light :slightly_smiling_face:

Thanks to read my post.

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How to change pound key # to direct call in grandstream ip phone

February 11, 2019, 11:46 pm
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@hunterman wrote:

Hello to the community,

I am pretty new in the freepbx community,
I have ip phone GRANDSTREAM model GXP2160 and connected to server voip “Asterisk now” all of call are good, My problem i need to active # “pound key” on the ip phone to make call like (#100) when press this #100 i need call to extension 100 , because already the # on the ip phone make redial call at the last call when i received.
Any help to change redial from # to make when press #100 to call direct to extension .

THANKS

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Problem with Asterisk server, when call from outside "GSM"

February 12, 2019, 3:46 am
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@hunterman wrote:

Dear Friends
My lab >>>
AM have GSM gateway device and connected with Asterisk server by sip trunk,
information of Asterisk server trunk:
Trunk Name: GSM
Outside:
type=peer
quality=yes
qualify=yes
host=x.x.x.x"private ip" for gsm gateway device

Incoming:

empty

I need to call from mobile number GSM to Asterisk server to ring directly to extension like 3000 in Asterisk server,
after applied configuration in GSM devices all was ok, because i tried to call from GSM to number of trunk to Asterisk server but extension no’t ringing, after that Am open ssh with Asterisk server and the result showing below:

[Feb 12 14:21:12] – Executing [3000@from-sip-external:1] NoOp("PJSIP/anonymous-00000014", "Received incoming SIP connection from unknown peer to 3000") in new stack

[Feb 12 14:21:12] – Executing [3000@from-sip-external:2] Set("PJSIP/anonymous-00000014", "DID=3000") in new stack

[Feb 12 14:21:12] – Executing [3000@from-sip-external:3] Goto("PJSIP/anonymous-00000014", "s,1") in new stack

[Feb 12 14:21:12] – Goto (from-sip-external,s,1)

[Feb 12 14:21:12] – Executing [s@from-sip-external:1] GotoIf("PJSIP/anonymous-00000014", "1?setlanguage:checkanon") in new stack

[Feb 12 14:21:12] – Goto (from-sip-external,s,2)

[Feb 12 14:21:12] – Executing [s@from-sip-external:2] Set("PJSIP/anonymous-00000014", "CHANNEL(language)=en") in new stack

[Feb 12 14:21:12] – Executing [s@from-sip-external:3] GotoIf("PJSIP/anonymous-00000014", "1?noanonymous") in new stack

[Feb 12 14:21:12] – Goto (from-sip-external,s,5)

[Feb 12 14:21:12] – Executing [s@from-sip-external:5] Set("PJSIP/anonymous-00000014", "TIMEOUT(absolute)=15") in new stack

[Feb 12 14:21:12] – Channel will hangup at 2019-02-12 14:21:27.691 +03.

[2019-02-12 14:21:12] WARNING[10608][C-0000004b]: func_channel.c:463 func_channel_read: Unknown or unavailable item requested: ‘recvip’

[Feb 12 14:21:12] – Executing [s@from-sip-external:6] Log("PJSIP/anonymous-00000014", "WARNING,"Rejecting unknown SIP connection from "") in new stack

[2019-02-12 14:21:12] WARNING[10608][C-0000004b]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "

[Feb 12 14:21:12] – Executing [s@from-sip-external:7] Answer("PJSIP/anonymous-00000014", "") in new stack

[Feb 12 14:21:12] > 0x7f63fc008380 – Strict RTP learning after remote address set to: 192.168.33.169:8012

[Feb 12 14:21:13] > 0x7f63fc008380 – Strict RTP switching to RTP target address 192.168.33.169:8012 as source

[Feb 12 14:21:13] – Executing [s@from-sip-external:8] Wait("PJSIP/anonymous-00000014", "2") in new stack

[2019-02-12 14:21:13] WARNING[10608][C-0000004b]: channel.c:5600 set_format: Unable to find a codec translation path: (slin) -> (g723)

[2019-02-12 14:21:13] ERROR[10608][C-0000004b]: channel.c:8073 ast_channel_start_silence_generator: Could not set write format to SLINEAR

[Feb 12 14:21:15] – Executing [s@from-sip-external:9] Playback("PJSIP/anonymous-00000014", "ss-noservice") in new stack

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: channel.c:5600 set_format: Unable to find a codec translation path: (slin16|g722|alaw|ulaw) -> (g723)

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: file.c:1245 ast_streamfile: Unable to open ss-noservice (format (g723)): Function not implemented

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: app_playback.c:492 playback_exec: Playback failed on PJSIP/anonymous-00000014 for ss-noservice

[Feb 12 14:21:15] – Executing [s@from-sip-external:10] PlayTones("PJSIP/anonymous-00000014", "congestion") in new stack

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: channel.c:5600 set_format: Unable to find a codec translation path: (slin) -> (g723)

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: indications.c:140 playtones_alloc: Unable to set ‘PJSIP/anonymous-00000014’ to signed linear format (write)

[2019-02-12 14:21:15] NOTICE[10608][C-0000004b]: app_playtones.c:98 handle_playtones: Unable to start playtones

[Feb 12 14:21:15] == Spawn extension (from-sip-external, s, 10) exited non-zero on ‘PJSIP/anonymous-00000014’

[Feb 12 14:21:15] – Executing [h@from-sip-external:1] Hangup("PJSIP/anonymous-00000014", "") in new stack

[Feb 12 14:21:15] == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00000014’

AsteriskNOW*CLI>

Any help to solve the problem ?
THANKS

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How to route all calls through 1 nic e.g eth1

February 12, 2019, 5:39 am
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@Arsalan wrote:

hello, I got freepbx 13.0.192.16-1.shmz65.1.57 and with 2 nic configured on eth0 and eth3 but for some reason freepbx is using eth3 for outgoing calls and if it got disconnected from eth3 trunk stops working …so what I need is if some one can help me and let me know how to configure it on another nic or reroute all calls on eth0 if eth3 fails or if I can configure it on eth2 for external dialing.

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Quick beep sound before ring-back on internal calls

February 12, 2019, 10:23 am
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@mcharlebois wrote:

Using FreePBX 14.0.5.25

We get a quick beep before the ring-back, but only on Internal calls.

Did some PCAP capture on both internal and external call and only major difference I see is that on external TRUNK calls my Calix ATA only receives a “183 Session Progress” response with no “180 Ringing” while during the internal call setup I see two “180 Ringing” response and no “183 Session Progress”

Only reference I found to this issue is the following but it hasn’t helped me much…
http://wiki.snom.com/FAQ/Why_do_I_hear_a_beep_before_ringing_sound_when_making_call

I have tried removing the “r” from the Asterisk dial options but it hasn’t changed anything.

Any help or insight would be greatly appreciated.

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FreePBX & Ubiquiti Edgemax Router Lite

February 12, 2019, 12:33 pm
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@MC1 wrote:

Hello

I realize this question might be more appropriate in a Ubiquiti Forum, however I thought I’d post something here in case someone has had a similar experience.

I’m looking for help is setting up my EdgeRouter Lite to work with FreePBX. Specifically what configuration on the Edge Router Lite will allow username & password authenticated trunk to work / register properly with my SIP provider. I don’t think I require port forwarding however do I need to put the FreePBX into a DMZ and if so how and I prefer to use the GUI to do this? I do have “eth2” available on the router if that helps. Also does anyone know what role the FreePBX firewall will play in the set up? Currently my SIP trunk is showing as registered with my provider, however at their end they keep seeing it register then un-register. Recently I defaulted the Edge Router and used the “Wan2Lan” setup wizard to configure the device.

Any information or links to video’s / documentation would be greatly appreciated. Thanks in advance for your help.

Michael

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List Pick Group members

February 12, 2019, 1:42 pm
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Remove HA and start using warm spare

February 12, 2019, 6:43 pm
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@bajramia wrote:

I have HA configured on PBXact 13.0.66 I have 1200 extentsion 600 Voip and 600 analog
Im using Vega 3050 for analog gateways i would like to ise the module Vega but only is supported on PBXact 14. Also i know HA foe 14 is close to be released so, is there anyway i can stop / destroy the cluster without reinstalling everything.

Thank you

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What's the all steps if i need to call from number mobile by gsm to Asterisk server

February 13, 2019, 2:14 am
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@hunterman wrote:

Hello everyone
In my office i need to make call from gsm mobile to direct extension number like 300,
my information about lab:
. GSM device
. Asterisk server

  • extension, sip like 300
  • Applied trunk between asterisk server and gsm gateway so the statutes between them is ok.
    note :
    Am can make call from asterisk extension “300” to call direct to gsm mobile by used (prefix and matched pattern), and finally worked as fine without any problem.

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CHAN_SIP technology listening on Port 5160

February 13, 2019, 5:08 am
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@hunterman wrote:

Hello everyone

When Am opened the extension like 100, I saw in General :

“This device uses CHAN_SIP technology listening on Port 5160 (UDP - this is a NON STANDARD port)”

Any one can told me whats meaning that?

Or how can i change port of sip from 5160 like ip server 192.168.1.10:5160 to port 5060 like 192.168.1.10:5060 when Am using SIP, And now using PJSIP, because my office infrastructure and all devices worked with port 5060 SIP, Can i change it to goal the compatibility between all devices.

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Vega 3050 to IP Phone

February 13, 2019, 6:33 am
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@bajramia wrote:

Hi All,
I have PBXact set up with Vega 3050 i can call from IP phone to analog but when i call from analog to Ip phone i get this message
chan_sip.c:26363 handle_request_invite: Failed to authenticate device “1A 12B” sip:6120@192.168.25.10;tag=00F6-000E-C89B1497
gateway are set registration mode FXO ports What im missing thank you.

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Vega50 "No Channel Available"

February 13, 2019, 8:12 am
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@decibel83 wrote:

Hi,

I have a Sangoma Vega50 connected to 2 ISDN lines, so 4 concurrent total calls.

We are using it with FreePBX.

Both inbound and outbound calls works good, but we can use only one channel per BRI. For example:

  1. I make an inbound call to the BRI1 number, IVR answers
  2. I make an inbound call to the BRI2 number, IVR answers
  3. I try to make an outbound call to any external number, I get congestion

I enabled debug on Vega50 (I am attaching the entire log during the 3 concurrent calls), and the third call is getting “SIP/2.0 500 No Channel Available” error. So it seems that only one channel is working on each BRI.

Could you help me please?

Thank you very much.

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3CX to FreePBX migration help needed

February 13, 2019, 8:57 am
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@DennisT wrote:

We’re in the process of migrating from 3CX to FreePBX. The consultant I hired to do this started the migration and completed some of it but has now gone silent. I need someone to complete the project.

FreePBX (current) is on a VM. We have approx 50 extensions consisting mostly of Yealink phones. SNOM PA1 paging, Patton SN4114/JS/EUI for 4 analog devices (fax, postage, etc), Patton SN4970 (PSTN T1), 24 DID, etc.

I addition we would be interested in support as needed after the migration. This would include a move to a SIP provider (voip.ms or equiv) after the 3CX to FreePBX migration is done.

PM me with experience, rates, etc. Thanks

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Pass DID Between 2 FreePBX server

February 13, 2019, 9:32 am
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@fgunno wrote:

I hope someone can help me with that:
My Situation:
I have 2 FreePBX server,
FreePBX1 have a trunk from our provider and receive the call first, FreePBX1 have a couple of extension configured.
FreePBX2 have a trunk from FreePBX1 (one of the extension of the FreePBX1)
Inbound route of FreePBX1 to Extension linked to FreePBX2 and everything work fine.
But Now I need to setup 2 Inbound route on FreePBX2, and the problem is that the DID is not passed thru FreePBX1 to FreePBX2 so the DID Specific inbound route is not working on FreePBX2.
I make it work with a Set Caller ID on FreePBX1 and a CID Inbound Route on FreePBX2, but the problem is the redial or notifications on the phone show the CID number and don’t show the real Caller ID.

What I want is to pass the DID from FreePBX1 to FreePBX2 directly. So Inbound Route on FreePBX2 will work as expected.

Thanks!

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Polycom phone goes Unreachable

February 13, 2019, 12:28 pm
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@brianjohnson wrote:

We have a number of Polycom phones at a remote site and I am seeing ONE phone become Unreachable periodically throughout the day. There are multiple phones of the same model but only this one seems to be a problem child while others seem to stay connected.

We are using PJSIP and I have done logging. I see the SIP OPTIONS packet sent and responded to every minute and then BAM the phone just stops responding. I have to think firewall config and internet connection are OK since other phones at the same site continue to respond correctly during the same time intervals and appear to work fine making/receiving calls.

I have been able to identify one thing which so far seems consistent when the phone stops responding. It looks like the phone was always on a call that was terminated less than a minute prior to when it stops responding to OPTIONS commands. From the PBX side it looks like it goes through normal call hangup but maybe the phone doesn’t know the call has been terminated?

Anyone seen this behavior before? This is on FreePBX 14.0.5.2. And yes, we using older Polycom phones (430’s). The phones could be upgraded if this is the problem but I am looking for some concrete explanation before I start doing random things to address.

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Asterisk internal DB and Freepbx GUI

February 13, 2019, 10:11 pm
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@oleggram wrote:

I need to change remotely (w/o GUI) Default User for particular Device in Deviceandusers mode. I can do it via internal Asterisk DB - Database put AMPUSER MyExtension/device NewDevice. It is working correctly with new combination but w/o any changes in GUI. What I need to do in order to correct info in GUI? Some type of synchronisation between internal Asterisk DB and FreePBX GUI ? Application/ Devices/Default User field is different from AMPUSER/MyExtension/device field in Database.

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