@tanvirkdi wrote:

Hello All,

we have a client that has freepbx v12 installed on their network. They have over 2500 extensions and apply configs can take between 4-5 mins which is fine. As this customer grows they are adding in additional users to help maintain their PBX systems. Is there a way to only restrict one user to Apply Config at a time after making changes?

The situation we are running into is that user1 has an extension open, and user2 has an extension open. If user1 and user2 make a change in freepbx, they are both presented with the Apply Config button at the top of their browser. Lets say that user1 walks away for a min and in that time, user2 hits apply config. during that time user1 walks back up and applys config. Is there a way of locking out the apply config if someone is already applying it to the system?

thanks

Posts: 1

Participants: 1

Read full topic

@adtopkek wrote:

Whenever I dial *21nnn from a different extension to toggle another extension’s follow me it just toggles the calling extension’s follow me and leaves the other extensions’s follow me how it was set. Is anyone else having issues with this?

I’m using the Freepbx 13 distro and I have the follow me app fully updated.

Posts: 3

Participants: 2

Read full topic

@CPORich wrote:

Title is pretty self explanatory

Looking for opinions on how to best configure ECM as there are multiple points to turn it on and off.

If it is of any help:
FreePBX 14.0.1.24
Asterisk 13.18.4
Cisco SPA112 ATA adapter
HP M426fdw printer/fax machine
flowroute trunk

T.38 pass through is setup and functioning

Posts: 4

Participants: 2

Read full topic

@CyberAndy wrote:

Dear all,

tried to install WebRTC, but getting a strange error message:
(webrtc/Webrtc.class.php:62) - Unsupported Version of Asterisk, You need at least 11.11 you have 15.1.4

Is there a bug in the version check?

Regards,
Andreas

Posts: 1

Participants: 1

Read full topic

@mcarnahan wrote:

I attempted to upgrade from FreePBX 13 to 14 on my Sangoma Appliance PBX using the 13 to 14 upgrader Module. It stated that since this was an official distro I had to update using the CLI and gave Upgrading from FreePBX 10.13.66 to SNG7 from the wiki for the instructions. I ran the RPM command: yum -y install (Ommitted URL As I’m Not allowed)

Initially I ran ‘distro-upgrade’ and it said that some of the dependencies needed to be upgraded, so I ran the yum update with completed with no issues. So I ran ‘distro-upgrade’ again. This time it reported that some of the packages installed were 32-bit and needed to be uninstalled and reinstalled as 64-bit. That completed with no issue.

Next I ran ‘distro-upgrade’ again and it completed, prompting me to "reboot the appliance at my convenience. "
I issued the reboot command and when it booted back up, I was expecting the green grub menu screen with the system upgrade option, but instead got a blue screen with two options:

SHMZ (2.6.32-642.6.2.el6.x86_64)
SHMZ 6 (2.6.32-504.8.1.el6.x86_64)

The 4 second timer ran out and the first option ran and resulted in this:

sd 1:0:0:0: [sda] 234441648 512-byte logical blocks: (120 GB/111 GiB)
sd 1:0:0:0: [sda] Write Protect is off
sd 1:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA
 sda: sda1 sda2 sda3
sd 1:0:0:0: [sda] Attached SCSI disk
usb 1-2: new high speed USB device number 2 using xhci_hcd
EXT4-fs (sda2): mounted filesystem with ordered data mode. Opts:
dracut: Mounted root filesystem /dev/sda2
usb 1-2: New USB device found, idVendor=05e3, idProduct=0608
usb 1-2: New USB device strings: Mfr=0, Product=1, SerialNumber=0
usb 1-2: Product: USB2.0 Hub
usb 1-2: configuration #1 chosen from 1 choice
hub 1-2:1.0: USB hub found
hub 1-2:1.0: 4 ports detected
dracut: chroot: failed to run command `/sbin/load_policy': Input/output error
dracut: Switching root
Kernel panic - not syncing: Attempted to kill init!
Pid: 1, comm: switch_root Not tainted 2.6.32-642.6.2.el6.x86_64 #1
Call Trace:
 [<ffffffff815482b1>] ? panic+0xa7/0x179
 [<ffffffff8112aea0>] ? perf_event_exit_task+0xc0/0x340
 [<ffffffff81081f97>] ? do_exit+0x867/0x870
 [<ffffffff81081ff8>] ? do_group_exit+0x58/0xd0
 [<ffffffff81082087>] ? sys_exit_group+0x17/0x20
 [<ffffffff8100b0d2>] ? system_call_fastpath+0x16/0x1b

Next I rebooted and selected the second option and got this:

sd 1:0:0:0: [sda] 234441648 512-byte logical blocks: (120 GB/111 GiB)
sd 1:0:0:0: [sda] Write Protect is off
sd 1:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA
 sda: sda1 sda2 sda3
sd 1:0:0:0: [sda] Attached SCSI disk
usb 1-2: new high speed USB device number 2 using xhci_hcd
EXT4-fs (sda2):
usb 1-2: New USB device found, idVendor=05e3, idProduct=0608
usb 1-2: New USB device strings: Mfr=0, Product=1, SerialNumber=0
usb 1-2: Product: USB2.0 Hub
usb 1-2: configuration #1 chosen from 1 choice
hub 1-2:1.0: USB hub found
hub 1-2:1.0: 4 ports detected
mounted filesystem with ordered data mode. Opts:
dracut: Mounted root filesystem /dev/sda2
dracut: chroot: failed to run command `/sbin/load_policy': Input/output error
dracut: Switching root
Kernel panic - not syncing: Attempted to kill init!
Pid: 1, comm: switch_root Not tainted 2.6.32-504.8.1.el6.x86_64 #1
Call Trace:
  [<ffffffff815292d6>] ? panic+0xa7/0x16f
  [<ffffffff8107a5f2>] ? do_exit+0x862/0x870
  [<ffffffff8107a658>] ? do_group_exit+0x58/0xd0
  [<ffffffff8107a6e7>] ? sys_exit_group+0x17/0x20
  [<ffffffff8100b072>] ? system_call_fastpath+0x16/0x1b
drm_kms_helper: panic occurred, switching back to text console

As of right now I have a none functioning system. I do have a backup, but don’t really want to reinstall if there is another fix.

Any help would be greatly appreciated!! Thanks

Posts: 1

Participants: 1

Read full topic

@assos40 wrote:

Hi,
I am trying to backup the cron directory but i get
"rsync: change_dir “/var/spool/cron” failed: Permission denied (13)
rsync error: some files/attrs were not transferred (see previous errors) (code 23) at main.c(1052) [sender=3.0.9]"
Any ideas
Thanks
Edit: I alrady tried fwconsole chown with no success

Posts: 4

Participants: 2

Read full topic

@steve_pbuk wrote:

Hi

I’m currently setting up a FreePBX server with the use of Sangoma S500 phones for the users.

Almost everything is working as expected but I have one slight issue.

I have an incoming route that does the following:

1. Played a recording (welcome message advising call will be recorded, etc)
2. Rings a ring group (8002) with hunt setup
   2.a Rings the reception phone for 20 seconds.
   2.b If no answer keeps ringing the reception phone and now rings the Sales ring group for 20 seconds
   2.c If still not answer keep ringing the above phones and now ring the Aftersales ring group for another 20 seconds.
3. As part of ring group 8002 if no one answers the “Destination if no answer” is set to terminate the call with the option of busy.

The problem I have is the external caller receives the busy tone and the call drops for them. But the reception phone keeps ringing with the external users phone number on the display. All the other phones in Sales and Aftersales stop ringing.

If I change the Destination if no answer to “Hangup” that works as expected but I don’t want to use this because its a bit “harsh” to the external caller.

If I try “busy” or “congestion” I get the problem of the reception phone still ringing after the call has ended.

System is uptodate with the latest patches and module updates:

Current PBX Version:14.0.1.24
Current System Version:12.7.4-1712-2.sng7

Current Asterisk Version: 13.18.4

Posts: 1

Participants: 1

Read full topic

@mailrishigupta wrote:

Hi all

I am a newbie to using FreePBX and need your help with a small problem regarding integrating my FreePBX setup with the traditional Electronic PABX (6 pots trunks, 16 analog extensions).

My current setup at home/small office involves:

1. Electronic PABX (6 pots trunks, 16 analog extensions)
2. FreePBX server running on a spare PC
3. SPA3102 > connected to one of the EPABX analog extensions (number 29) on the FXO port, and connected via LAN to FreePBX server, where it is defined as a PJSIP trunk.

What I want to achieve is dial a POTS trunk from outside, and be able to transfer the call to one of the ANALOG extensions of my electronic PABX (This feature is not supported by my PABX at present, and is very costly to implement)

What I have achieved is the following:

1. I call one of the POTS trunks from outside
2. Electronic PABX rings that trunk on extension 29 (connected to SPA3000 FXO)
3. SPA3000 rings through to FreePBX
4. FreePBX auto-answers and responds with IVR
5. I am able to then dial whichever FREEPBX extension I want (not what I wanted to achieve).

Suppose I am on extension 21. To transfer an inbound call on my electronic PABX to another analog extension, I press Flash+extn no (Eg Flash+23). This will put inbound POTS trunk on hold, ring extension 23, and when I hang up extension 21, connect inbound POTS trunk to extn 23.

What I want is that after FreePBX answers and starts IVR, I dial the required EPABX extension, and then FreePBX does the following:

1. Dial Flash ON THE SAME INBOUND TRUNK FROM WHICH I AM CALLING
2. Dial the EPABX extension number I want
3. Hang up, so that my call gets transferred to the required analog extension (switching done by EPABX). At this point, FreePBX is completely out of the loop again.

I hope I have been able to explain my problem properly.

I look forward to your suggestions.

Regards

Posts: 1

Participants: 1

Read full topic

@roderick1st wrote:

Hello FreePBX community.

I have recently installed FreePBX and connected to Sipgatecouk BASIC account. After a lot of digging and learning I have successfully managed to make and receive calls from Sipgate using SIP_CHAN. Having read that PJSIP is the new way to do things I have been attempting to configure the sip trunk to work with PJSIP.

I can receive incoming calls fine, just as with CHAN_SIP but I can’t make out going calls. I have setup all settings the same (which can be the same) for both trunks. I think it has something to do with the connection string being sent to the trunk.

CHAN_SIP (Working)
– Called SIP/sipgatecouk/447700770077

PJSIP (Not working)
– Called PJSIP/907700770077@Sipgate_Trunk_PJSIP
I am assuming if I can get this string to read Called PJSIP/sipgatecouk/447700770077 it would work.

It seems to be ignoring my Dial Pattern with the only one I have at the moment is in the outbound route and is ()9|[./]

Any help would be much appreciated. Full log for PJSIP failed call below.

 to Asterisk 13.18.4 currently running on pbx (pid = 1905)
  == Setting global variable 'SIPDOMAIN' to 'pbxmydomaincom'
    -- Executing [907700770077@from-internal:1] Macro("PJSIP/101-00000009", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/101-00000009", "TOUCH_MONITOR=1514829635.14") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/101-00000009", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/101-00000009", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/101-00000009", "AMPUSERCIDNAME=Office") in new stack
    -- Executing [s@macro-user-callerid:8] ExecIf("PJSIP/101-00000009", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("PJSIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/101-00000009", "AMPUSERCID=101") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/101-00000009", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:12] Set("PJSIP/101-00000009", "CALLERID(all)="Office" <101>") in new stack
[2018-01-01 18:00:35] WARNING[14983][C-00000007]: ast_expr2.fl:470 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '>', expecting '-' or '!' or '(' or '<token>'; Input:
"LIMIT"="LIMIT" & 3 & 1 & >0 & 0>=
                          ^
[2018-01-01 18:00:35] WARNING[14983][C-00000007]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
    -- Executing [s@macro-user-callerid:13] GotoIf("PJSIP/101-00000009", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:14] ExecIf("PJSIP/101-00000009", "1?Set(GROUP(concurrency_limit)=101)") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/101-00000009", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:16] NoOp("PJSIP/101-00000009", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("PJSIP/101-00000009", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] GotoIf("PJSIP/101-00000009", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,37)
    -- Executing [s@macro-user-callerid:37] Set("PJSIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [s@macro-user-callerid:38] Set("PJSIP/101-00000009", "CALLERID(name)=Office") in new stack
    -- Executing [s@macro-user-callerid:39] GotoIf("PJSIP/101-00000009", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:40] Set("PJSIP/101-00000009", "CDR(cnam)=Office") in new stack
    -- Executing [s@macro-user-callerid:41] Set("PJSIP/101-00000009", "CDR(cnum)=101") in new stack
    -- Executing [s@macro-user-callerid:42] Set("PJSIP/101-00000009", "CHANNEL(language)=en_GB") in new stack
    -- Executing [907700770077@from-internal:2] Gosub("PJSIP/101-00000009", "sub-record-check,s,1(out,907700770077,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("PJSIP/101-00000009", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("PJSIP/101-00000009", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("PJSIP/101-00000009", "NOW=1514829635") in new stack
    -- Executing [s@sub-record-check:4] Set("PJSIP/101-00000009", "__DAY=01") in new stack
    -- Executing [s@sub-record-check:5] Set("PJSIP/101-00000009", "__MONTH=01") in new stack
    -- Executing [s@sub-record-check:6] Set("PJSIP/101-00000009", "__YEAR=2018") in new stack
    -- Executing [s@sub-record-check:7] Set("PJSIP/101-00000009", "__TIMESTR=20180101-180035") in new stack
    -- Executing [s@sub-record-check:8] Set("PJSIP/101-00000009", "__FROMEXTEN=101") in new stack
    -- Executing [s@sub-record-check:9] Set("PJSIP/101-00000009", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("PJSIP/101-00000009", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("PJSIP/101-00000009", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("PJSIP/101-00000009", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("PJSIP/101-00000009", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("PJSIP/101-00000009", "3?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("PJSIP/101-00000009", "1?sub-record-check,out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] NoOp("PJSIP/101-00000009", "Outbound Recording Check from 101 to 907700770077") in new stack
    -- Executing [out@sub-record-check:2] Set("PJSIP/101-00000009", "RECMODE=dontcare") in new stack
    -- Executing [out@sub-record-check:3] ExecIf("PJSIP/101-00000009", "1?Goto(routewins)") in new stack
    -- Goto (sub-record-check,out,7)
    -- Executing [out@sub-record-check:7] Gosub("PJSIP/101-00000009", "recordcheck,1(dontcare,out,907700770077)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/101-00000009", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/101-00000009", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("PJSIP/101-00000009", "") in new stack
    -- Executing [out@sub-record-check:8] Return("PJSIP/101-00000009", "") in new stack
    -- Executing [907700770077@from-internal:3] ExecIf("PJSIP/101-00000009", "0 ?Set(CDR(accountcode)=)") in new stack
    -- Executing [907700770077@from-internal:4] Set("PJSIP/101-00000009", "EMERGENCYROUTE=YES") in new stack
    -- Executing [907700770077@from-internal:5] Set("PJSIP/101-00000009", "MOHCLASS=default") in new stack
    -- Executing [907700770077@from-internal:6] ExecIf("PJSIP/101-00000009", "1?Set(TRUNKCIDOVERRIDE=01234567890)") in new stack
    -- Executing [907700770077@from-internal:7] Set("PJSIP/101-00000009", "_NODEST=") in new stack
    -- Executing [907700770077@from-internal:8] Macro("PJSIP/101-00000009", "dialout-trunk,1,907700770077,,on") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("PJSIP/101-00000009", "DIAL_TRUNK=1") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("PJSIP/101-00000009", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] ExecIf("PJSIP/101-00000009", "0?Set(CALLERID(num)=101)") in new stack
    -- Executing [s@macro-dialout-trunk:4] GotoIf("PJSIP/101-00000009", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("PJSIP/101-00000009", "DIAL_NUMBER=907700770077") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("PJSIP/101-00000009", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-dialout-trunk:7] Set("PJSIP/101-00000009", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [s@macro-dialout-trunk:8] Set("PJSIP/101-00000009", "DIAL_TRUNK_OPTIONS=T") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("PJSIP/101-00000009", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,11)
    -- Executing [s@macro-dialout-trunk:11] GotoIf("PJSIP/101-00000009", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:12] Macro("PJSIP/101-00000009", "outbound-callerid,1") in new stack
    -- Executing [s@macro-outbound-callerid:1] NoOp("PJSIP/101-00000009", "101") in new stack
    -- Executing [s@macro-outbound-callerid:2] NoOp("PJSIP/101-00000009", "") in new stack
    -- Executing [s@macro-outbound-callerid:3] NoOp("PJSIP/101-00000009", "off") in new stack
    -- Executing [s@macro-outbound-callerid:4] ExecIf("PJSIP/101-00000009", "0?Set(CALLERPRES(name-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:5] ExecIf("PJSIP/101-00000009", "0?Set(CALLERPRES(num-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:6] ExecIf("PJSIP/101-00000009", "0?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:7] GotoIf("PJSIP/101-00000009", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing [s@macro-outbound-callerid:11] Set("PJSIP/101-00000009", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:12] Set("PJSIP/101-00000009", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:13] Set("PJSIP/101-00000009", "TRUNKOUTCID=01234567890") in new stack
    -- Executing [s@macro-outbound-callerid:14] GotoIf("PJSIP/101-00000009", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,19)
    -- Executing [s@macro-outbound-callerid:19] ExecIf("PJSIP/101-00000009", "1?Set(CALLERID(all)=01234567890)") in new stack
    -- Executing [s@macro-outbound-callerid:20] ExecIf("PJSIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:21] ExecIf("PJSIP/101-00000009", "1?Set(CALLERID(all)=01234567890)") in new stack
    -- Executing [s@macro-outbound-callerid:22] ExecIf("PJSIP/101-00000009", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:23] ExecIf("PJSIP/101-00000009", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:24] Set("PJSIP/101-00000009", "CDR(outbound_cnum)=01234567890") in new stack
    -- Executing [s@macro-outbound-callerid:25] Set("PJSIP/101-00000009", "CDR(outbound_cnam)=") in new stack
    --
-- Executing [s@sub-flp-1:1] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=07700770077)") in new stack
    -- Executing [s@sub-flp-1:2] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:3] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=07700770077)") in new stack
    -- Executing [s@sub-flp-1:4] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:5] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=07700770077)") in new stack
    -- Executing [s@sub-flp-1:6] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:7] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=07700770077)") in new stack
    -- Executing [s@sub-flp-1:8] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:9] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=07700770077)") in new stack
    -- Executing [s@sub-flp-1:10] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:11] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=07700770077)") in new stack
    -- Executing [s@sub-flp-1:12] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:13] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=4407700770077)") in new stack
    -- Executing [s@sub-flp-1:14] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:15] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=4407700770077)") in new stack
    -- Executing [s@sub-flp-1:16] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:17] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=747608470)") in new stack
    -- Executing [s@sub-flp-1:18] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:19] ExecIf("PJSIP/101-00000009", "0?Set(TARGET_FLP_1=07700770077)") in new stack
    -- Executing [s@sub-flp-1:20] GotoIf("PJSIP/101-00000009", "0?match") in new stack
    -- Executing [s@sub-flp-1:21] Return("PJSIP/101-00000009", "") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("PJSIP/101-00000009", "OUTNUM=907700770077") in new stack
    -- Executing [s@macro-dialout-trunk:15] Set("PJSIP/101-00000009", "custom=PJSIP") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("PJSIP/101-00000009", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
    -- Executing [s@macro-dialout-trunk:17] ExecIf("PJSIP/101-00000009", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:18] Macro("PJSIP/101-00000009", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/101-00000009", "") in new stack
    -- Executing [s@macro-dialout-trunk:19] GotoIf("PJSIP/101-00000009", "0?skipcrm") in new stack
    -- Executing [s@macro-dialout-trunk:20] Set("PJSIP/101-00000009", "__CRM_DIRECTION=OUTBOUND") in new stack
    -- Executing [s@macro-dialout-trunk:21] Set("PJSIP/101-00000009", "__CRM_DESTINATION=907700770077") in new stack
    -- Executing [s@macro-dialout-trunk:22] Set("PJSIP/101-00000009", "__CRM_SOURCE=101") in new stack
    -- Executing [s@macro-dialout-trunk:23] AGI("PJSIP/101-00000009", "sangomacrm.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
    -- <PJSIP/101-00000009>AGI Script sangomacrm.agi completed, returning 0
    -- Executing [s@macro-dialout-trunk:24] Set("PJSIP/101-00000009", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:25] NoOp("PJSIP/101-00000009", "CRM Finished") in new stack
    -- Executing [s@macro-dialout-trunk:26] GotoIf("PJSIP/101-00000009", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:27] ExecIf("PJSIP/101-00000009", "1?Set(CONNECTEDLINE(num,i)=907700770077)") in new stack
    -- Executing [s@macro-dialout-trunk:28] ExecIf("PJSIP/101-00000009", "1?Set(CONNECTEDLINE(name,i)=CID:01234567890)") in new stack
    -- Executing [s@macro-dialout-trunk:29] ExecIf("PJSIP/101-00000009", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)01234567890)") in new stack
    -- Executing [s@macro-dialout-trunk:30] GotoIf("PJSIP/101-00000009", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:31] Dial("PJSIP/101-00000009", "PJSIP/907700770077@Sipgate_Trunk_PJSIP,300,T") in new stack
    -- Called PJSIP/907700770077@Sipgate_Trunk_PJSIP
 == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:32] NoOp("PJSIP/101-0000000a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:33] GotoIf("PJSIP/101-0000000a", "1?continue,1:s-CHANUNAVAIL,1") in new stack

Posts: 1

Participants: 1

Read full topic

@nsmith17044 wrote:

I have been trying to get outgoing calls to work on the GXW4104 but I can’t quite figure out the issue.

I have followed the WIKI article on configuration with FreePBX using a single trunk type structure (all FXOs work as a single trunk with FreePBX ). I’ve checked these settings a few times thinking I’ve missed something but all looks correct.

https://wiki.freepbx.org/display/FOP/Configuring+a+Grandstream+GXW-410X+Device+to+act+as+an+FXO+Gateway

I am using the CHAN_SIP driver on port 5060 as I need support for older Cisco SIP phones that PJSIP doesn’t seem support. I have observed too many issues trying to mix SIP and PJSIP devices so everything is using the CHAN_SIP driver.

When the outgoing INVITE from Asterisk is sent it looks like the GXW4101 is rejecting it (SIP 403). The SIP debug output is below the verbose monitoring.

Incoming calls route properly on all FXO ports for the GXW4104 with bi-directional audio verified.

In my TRUNK settings the trunk name matches the user ids in the 4101 and the outgoing settings are:
type=friend
qualify=yes
secret=mysecret
host=172.30.1.7
context=from-trunk
insecure=port
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw

The incoming fields are all BLANK per the wiki article.

I am beginning to suspect the SIP invite content that includes the %40GXW4104 characters is confusing the GWX4104 dialplan. However I don’t know how / where that is coming from. GXW4104 is the FREEPBX trunk name, not the SIP trunk name. I see in my SYSLOG file that the GXW4104 sees an invalid parsed dialplan length mesage and and the parsed number includes the 40GXW4104 part of the string.

QUESTION(s):
Any idea on what might be happening here?
Anybody have a better article than the Wiki article referenced on a FreePBX setup?
Any better way to troubleshoot this?

OUTPUT snippet:
– Executing [s@macro-dialout-trunk:23] Dial(“SIP/104-00000043”, “SIP/gxwt1/17174375277@GXW4104,300,Tt”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19732
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.30.1.7:5060:
INVITE sip:17174375277%40GXW4104@172.30.1.7 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.8:5060;branch=z9hG4bK2aaa6d7a
Max-Forwards: 70
From: sip:7172485553@172.30.1.8;tag=as6068fa0c
To: sip:17174375277%40GXW4104@172.30.1.7
Contact: sip:7172485553@172.30.1.8:5060
Call-ID: 31e0cb0c10375dee22f448bd1a0c64aa@172.30.1.8:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.7.1
Date: Mon, 01 Jan 2018 20:05:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245

v=0
o=root 155194336 155194336 IN IP4 172.30.1.8
s=Asterisk PBX 13.7.1
c=IN IP4 172.30.1.8
t=0 0
m=audio 19732 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

-- Called SIP/gxwt1/17174375277@GXW4104

<— SIP read from UDP:172.30.1.7:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.1.8:5060;branch=z9hG4bK2aaa6d7a
From: sip:7172485553@172.30.1.8;tag=as6068fa0c
To: sip:17174375277%40GXW4104@172.30.1.7
Call-ID: 31e0cb0c10375dee22f448bd1a0c64aa@172.30.1.8:5060
CSeq: 102 INVITE
User-Agent: Grandstream GXW4104 (HW 1.1, Ch:5) 1.4.1.5
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:172.30.1.7:5060 —>
SIP/2.0 403
Via: SIP/2.0/UDP 172.30.1.8:5060;branch=z9hG4bK2aaa6d7a
From: sip:7172485553@172.30.1.8;tag=as6068fa0c
To: sip:17174375277%40GXW4104@172.30.1.7;tag=9c474e62cbc16875
Call-ID: 31e0cb0c10375dee22f448bd1a0c64aa@172.30.1.8:5060
CSeq: 102 INVITE
User-Agent: Grandstream GXW4104 (HW 1.1, Ch:5) 1.4.1.5
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 172.30.1.7:5060:
ACK sip:17174375277%40GXW4104@172.30.1.7 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.8:5060;branch=z9hG4bK2aaa6d7a
Max-Forwards: 70
From: sip:7172485553@172.30.1.8;tag=as6068fa0c
To: sip:17174375277%40GXW4104@172.30.1.7;tag=9c474e62cbc16875
Contact: sip:7172485553@172.30.1.8:5060
Call-ID: 31e0cb0c10375dee22f448bd1a0c64aa@172.30.1.8:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.7.1
Content-Length: 0

---

[2018-01-01 15:05:30] WARNING[1888][C-00000020]: chan_sip.c:23372 handle_response_invite: Received response: “Forbidden” from 'sip:7172485553@172.30.1.8;tag=as6068fa0c’
Scheduling destruction of SIP dialog ‘31e0cb0c10375dee22f448bd1a0c64aa@172.30.1.8:5060’ in 6400 ms (Method: INVITE)

Posts: 2

Participants: 1

Read full topic

@Logistry wrote:

Happy new year everyone.

so i have my freepbx on my raspberry pi, using simonics as a trunk, i configured the trunk as described in their documentation (second method, peer)

outgoing calls work just fine, but some incoming calls are not being picked up by the ivr. i see in the logs the following

Executing [s@ivr-1:11] ExecIf("SIP/45.55.163.124-00000015", "1?Background(custom/main-newrec)") in new stack
    -- <SIP/45.55.163.124-00000015> Playing 'custom/main-newrec.slin' (language 'en')
       > 0x6e7f5338 -- Strict RTP switching to RTP target address 45.55.163.124:18468 as source
       > 0x6e7f5338 -- Strict RTP learning complete - Locking on source address 45.55.163.124:18468
  == Spawn extension (ivr-1, s, 11) exited non-zero on 'SIP/45.55.163.124-00000015'
    -- Executing [h@ivr-1:1] Hangup("SIP/45.55.163.124-00000015", "") in new stack
  == Spawn extension (ivr-1, h, 1) exited non-zero on 'SIP/45.55.163.124-00000015'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/45.55.163.124-00000015

i appreciate any help
thanks

Posts: 1

Participants: 1

Read full topic

@joshb wrote:

How can I apply the PITCH_SHIFT and VOLUME dialplan functions to all incoming calls from a SIP trunk?

Posts: 1

Participants: 1

Read full topic

@GlasairII wrote:

My first post after gaining much wisdom from this forum for some 5+ years.

My question is simply “Can the Warm Spare Setup communicate on another port other than 22?”.

I followed the Wiki which worked perfectly. I had previously changed the SSH port number so it wouldn’t get bott’ed all day but had to default them back to port 22 in order to get the warm spare to work. Is it possible to have them communicate on another port? Just didn’t know if it was something built into the code of FreePBX.

Thanks Gents

Posts: 1

Participants: 1

Read full topic

@kzip2008 wrote:

Hi All,

I was wondering if it was possible to lock down an admin group to only view/modify a set range of extensions ?

Thanks

Posts: 1

Participants: 1

Read full topic

@chriskinsey wrote:

I have roll-my-own distro installations of FreePBX 14/Asterisk 13 on AWS, Linode, Vultr (takes ISO direct from FPBX), and Digital Ocean. No problems except my Grandstream phones that worked under FPBX <14 needed reconfiguration. My issue. Glad to be on Centos 7 and nearing N or N-1 on Linux apps. Vanilla is good.

Fair warning: The following information will make perfect sense to super geeks, and should be avoided by FreePBX novices. It is extremely easy to render yourself inoperative with improper firewall rules.

The Responsive Firewall is great and has withstood hundreds of attacks from the usual countries and trolls without a single successful intrusion. I have left Chan_SIP enabled because my remote endpoints are behind variable telco IPs or softphones on roaming mobile phones. The latest hack hitting me has been sip-external attempts on the devices. None have been successful as I use oddball extension numbers and very complex passwords. However, I wanted to close this loophole without having to revert to No-IP (works great) or similar which just creates another point of failure in the system. Plus, allowing all those unsuccessful hacks is a bandwidth and processor drain.

My answer - turn off Chan_SIP in Responsive Firewall and go back to explicit permissions “by type.” Here is what I mean when I say “by type.” I noticed that I have never been hacked by a telco or mobile provider IPs. This is probably because it would be expensive - and traceable - to use mobile devices as hotspots for high volume devices running hundreds of thousands of transactions an hour. It is also traceable to use a telco IP address for such purposes. The question was, how to prevent the majority - if not all - bad guys while allowing responsible good guys to get into my system without leaving SIP ports open to the Internet at large.

There is a very handy Website https://whoer.net/checkwhois which provides very useful information when you input an IP address. For instance, when I put in the IP address currently assigned to my home, the site tells me the entire IP range to which my IP belongs. So, when I gave it ###.###.###.### it came back with ###.###.0.0/14. Now rather than relying on No-IP, I simply open up the entire 65K+ IPs which could possibly be assigned to me and remove that point of failure.

The Website above is very useful if you know the IP address - but what if you have no clue such as in the case of mobile providers? Now we resort to a Web site tracemyip.org and some geekery. This site will allow you to search an org name and show you all IPs which have been reported active recently. You can then take these individual IPs and scrape them into Excel or your favorite data sift and sort tool and use the Website above to get ranges. There are specific search terms to find each of the providers. I do not represent that these searches will provide an exhaustive list. To some extent this is up to your persistence and sleuthing. AT&T actually provides a listing of IPs for developers but I found that this list excludes most specific IPs which might be assigned to individual mobile phones and rather concentrates on infrastructure IPs. I used their list combined with the method above to develop my trusted list for AT&T Wireless. I used a posting in this forum plus the technique above to develop my Verizon Wireless list.

Remember, the data will not be handed to you on a plate by these sites. You must develop and refine your own. I just decided it was better to trust good, relatively safe people and lock down the world rather than continue refining blocklists. The load on my machines will be lower and the burden on me personally will almost go away.

I also highly recommend an app called Subnet Plus which allows you to fiddle around with blocks of IPs to reduce your firewall exceptions. It is available in the Apple App Store. Sorry Androids, I don’t know or care if it is available there! In any event, it is that app tool and a combination of the techniques above which allowed me to develop these rules.

Once I had entered these rules, I tested by REMOVING all of my No-IP hostnames from the firewall rules. I watched my softphones deregister and immediately reregister using the carrier rules instead of No-IP permissions. Calls completed about 2 seconds faster simply because a step (and more importantly a point of failure) had been removed from the equation. Once I confirmed my rules were working I turned off Chan_SIP in the Responsive Firewall, power cycled my mobile, rebooted my premise endpoints and…success! Take that Palestinian, Russian, Chinese, Yemeni, and other trolls from this weekend!

Your results may vary. All I know is I can now just look at Bria Mobile as a single point of failure and not have to ask users “Is your No-IP app running? What does it say? Refresh the IP. Now what does it say? OK now let’s go to Bria while I step in front of this oncoming bus…”

I can confirm that the Verizon rules work - that is my carrier. Here are the searches for the major mobile carriers:

Verizon Wireless
https://tools.tracemyip.org/search–isp/verizon+wireless

AT&T Wireless
https://tools.tracemyip.org/search–org/at%26t+wireless
The AT&T-provided list to which I refer above can be found here - remember this is not exhaustive:
https://developer.att.com/technical-library/network-technologies/ip-addresses

T-Mobile
https://tools.tracemyip.org/search–isp/T-Mobile+USA

Sprint PCS
https://tools.tracemyip.org/search–isp/Sprint+PCS:-v-:&gTr=51&gNr=50

And to get you started, here are several million IP addresses for Verizon and AT&T, just cut and paste into your console or paste into a bash file. If pasting into bash, remember to remove all non-fwconsole lines or comment text lines out. If you find when you are using a softphone on a mobile device that you can not get in do one of two things: a) check your phone to see what IP address is currently assigned, b) check your Responsive Firewall to see which IP is currently blocked and then use the instructions above to add a new block to the permitted list for your carrier. Remember that carriers like AT&T, Sprint, and Verizon also have landline and business services with separate IP ranges. You should not need to enter those ranges for mobile softphone usage.

You must have the commercial sysadmin module (only $25, just buy it - your time is worth more than that) for all the firewall features to work.

Happy New Year!

Verizon Wireless

fwconsole firewall add trusted 166.128.0.0/9
fwconsole firewall add trusted 174.192.0.0/10
fwconsole firewall add trusted 97.128.0.0/9
fwconsole firewall add trusted 70.192.0.0/11
fwconsole firewall add trusted 69.96.0.0/13
fwconsole firewall add trusted 69.82.0.0/15
fwconsole firewall add trusted 66.174.0.0/16
fwconsole firewall add trusted 72.96.0.0/11
fwconsole firewall add trusted 75.192.0.0/10
fwconsole firewall add trusted 97.0.0.0/10

ATT Wireless

Note: The ranges below are expanded beyond the above technical reference for coding simplicity (200 line reduction!)

All addresses are AT&T Wireless - not AT&T - addresses

fwconsole firewall add trusted 107.64.0.0/10
fwconsole firewall add trusted 160.170.220.0/22
fwconsole firewall add trusted 166.128.0.0/13
fwconsole firewall add trusted 166.136.0.0/15
fwconsole firewall add trusted 166.138.0.0/16
fwconsole firewall add trusted 166.147.104.0/25
fwconsole firewall add trusted 166.170.0.0/19
fwconsole firewall add trusted 166.170.32.0/20
fwconsole firewall add trusted 166.170.48.0/21
fwconsole firewall add trusted 166.170.56.0/22
fwconsole firewall add trusted 166.171.56.0/22
fwconsole firewall add trusted 166.171.120.0/22
fwconsole firewall add trusted 166.171.184.0/22
fwconsole firewall add trusted 166.171.248.0/22
fwconsole firewall add trusted 166.172.56.0/22
fwconsole firewall add trusted 166.172.60.0/22
fwconsole firewall add trusted 166.172.120.0/22
fwconsole firewall add trusted 166.172.184.0/22
fwconsole firewall add trusted 166.172.188.0/22
fwconsole firewall add trusted 166.173.56.0/22
fwconsole firewall add trusted 166.173.60.0/22
fwconsole firewall add trusted 166.173.184.0/22
fwconsole firewall add trusted 166.173.248.0/22
fwconsole firewall add trusted 166.175.56.0/22
fwconsole firewall add trusted 166.175.60.0/22
fwconsole firewall add trusted 166.175.184.0/22
fwconsole firewall add trusted 166.175.188.0/22
fwconsole firewall add trusted 166.176.56.0/22
fwconsole firewall add trusted 166.176.120.0/22
fwconsole firewall add trusted 166.176.184.0/22
fwconsole firewall add trusted 166.176.248.0/22
fwconsole firewall add trusted 166.177.56.0/22
fwconsole firewall add trusted 166.177.120.0/22
fwconsole firewall add trusted 166.177.184.0/22
fwconsole firewall add trusted 166.177.248.0/22
fwconsole firewall add trusted 166.216.133.103/32
fwconsole firewall add trusted 166.216.133.208/28
fwconsole firewall add trusted 166.216.133.231/32
fwconsole firewall add trusted 166.216.133.231/32
fwconsole firewall add trusted 166.216.133.64/28
fwconsole firewall add trusted 166.216.157.0/24
fwconsole firewall add trusted 166.216.158.0/24
fwconsole firewall add trusted 166.216.159.0/24
fwconsole firewall add trusted 166.216.165.0/24

Posts: 3

Participants: 2

Read full topic

@perrcla wrote:

Hi,

freepbx 10.13.66-22 with some commercial module.

The statistic widget show a peak with 69 active call… non realistic in my enviroment…
Usually i have max 10 concurrent calls.

What should I investigate?

Posts: 3

Participants: 2

Read full topic

@mvogel4949 wrote:

I no longer see the sangoma property management module in the edge release. Has it been pulled back?

Posts: 7

Participants: 2

Read full topic

@mannebk wrote:

Hi Folks,

just in case somebody has similar problems:

if you run businesse and privat accounts with easybell Germany, they expect the outgoing CID to be like this:

examples for Germany:

<countrycode including leeding zeros followed by areacode w/o zero and number>
<00497115555555>

while the business account also accepts the local version
<07115555555>

the privat account does not and but accepts a mixture, the international w/o leading zeros
<497115555555>

took me quite some time to figguer that one out. It was a lucky hit. And easybell was a bit helpfull too. they have a very good customer support there. the support now investigates why its different on different account typs, and they will fix or update their fqa pages.

so if they fix this by kicking out everything but the 0049 version, and you dont have that in your pbx settings, you might discover your outgoing calls to not work any more in near future sorry for that then.

regards Manne

Posts: 1

Participants: 1

Read full topic

@waterwheel wrote:

I recently rearranged some of my phone lines (deleted some numbers, including our former outbound route), and now my outbound routes are not working.

I suspect either my outbound dial patterns (defaults generated by the system) or some way the dial patterns interface with my provider are to blame.

If I remove all dial patters, system just goes silent on outbound calls. When i use the system generated 7/10/11 digit dial patterns, when I dial a number like 15555555555, I get a busy signal.
Output from asterisk -RvvvvT^c command

Blockquote
[2018-01-02 11:59:45] WARNING[27619][C-00001b23]: app_macro.c:310 _macro_exec: No such context ‘macro-outisbusy’ for macro ‘outisbusy’. Was called by 1@from-internal
[Jan 2 11:59:45] – Executing [1@from-internal:6] Wait(“SIP/1-0000004b”, “1”) in new stack
[Jan 2 11:59:46] – Executing [1@from-internal:7] Congestion(“SIP/1-0000004b”, “20”) in new stack
[2018-01-02 11:59:46] WARNING[27619][C-00001b23]: channel.c:4883 ast_prod: Prodding channel ‘SIP/1-0000004b’ failed
[Jan 2 11:59:46] == Spawn extension (from-internal, 151, 7) exited non-zero on ‘SIP/1-0000004b’
[Jan 2 11:59:46] – Executing [h@from-internal:1] Macro(“SIP/1-0000004b”, “hangupcall”) in new stack
[Jan 2 11:59:46] – Executing [s@macro-hangupcall:1] GotoIf(“SIP/1-0000004b”, “1?theend”) in new stack
[Jan 2 11:59:46] – Goto (macro-hangupcall,s,3)
[Jan 2 11:59:46] – Executing [s@macro-hangupcall:3] ExecIf(“SIP/1-0000004b”, “0?Set(CDR(recordingfile)=)”) in new stack
[Jan 2 11:59:46] – Executing [s@macro-hangupcall:4] Hangup(“SIP/1-0000004b”, “”) in new stack
[Jan 2 11:59:46] == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/1-0000004b’ in macro ‘hangupcall’
[Jan 2 11:59:46] == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1-0000004b’
[Jan 2 11:59:46] == Extension Changed 1[ext-local] new state Idle for Notify User 2

Blockquote

Any ideas on how to diagnose /correct?

Posts: 2

Participants: 2

Read full topic

@Cartman wrote:

Very new to this. Using 1.6.2-7 and our GSM router is no longer working. I want to reroute calls to a different trunk/route. I have copied the dialing pattern and pasted it into the route Id like to use, changed the original route to the one Id like to use. What other steps would I need? This is the first time Ive used it, thank you.

Posts: 2

Participants: 2

Read full topic