@sknight06 wrote:

Hey all,

What is the best way to do a ring group which hunts through a few phones and then ends up at the voicemail box of the first extension in the group? Or is this possible? The problem is that I would like each extension in the group to have its own voicemail box but the caller gets stopped by voicemail while running through the extensions in the group.

Ideally, calls to an individual extension rather than a ring group would go to the voicemail box for that extension, while calls to the ring group would bypass the voicemail boxes until the very end when the ring group is done (no one answered).

This seems like a common thing to do in office settings, so I'm probably missing something simple.

Any help is very much appreciated!

Spence

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@Effaceurs wrote:

Hello everybody!
I have next configuration, analog PBX, PSTN Gateway Grandstream GXW410X, FreePBX server, analog phones and VOIP Phones.
I manage system in next way, from VOIP phone i be able to make a call to special extension (2000) and get into Gateway GXW410X, so after i hear beep and i dial internal number analog phones and speak with abonents.
I have the next question, when i recieve a call on my VOIP phone, i would like to redirect this call to internal analog extension, but i could only redirect to (2000). So how i can manage system and bind some extensions or something else, that they be able to call to extension 2000 and after beep, dial internal analog extension in automatic mode? For example i dial 2005 - and literally i dial to 2000 and after beep to 106 (internal extension) automatic. Hope you understand me.
Thank you.

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@lubo_petrov wrote:

I'm renting my apartment on airbnb.com My place is in multistory building. The only way for my future guest to enter is with key or by dialing code on the door intercom. The intercom is set to dial out to number that is registered with my freepbx. I was wondering if there is a way to accept this call and then to invoke freepbx to dial 9 in order to open door.

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@intravista wrote:

HI There,

I'm trying to setup a few different DID's for different businesses to ring on one phone. The phone is a cisco 7960.

I setup each DID with an inbound route to go to an extension.

I then used endpoint manager to map the extensions to the buttons...

So far so good... Each DID rings to the right extention (as in I get the right voicemail)... But the line doesn't ring.

Now, if I use the extension, I can make outbound calls, the DID is correct. All is good.

Now also, I looked at endpoint manager...

• The first extension shows the IP address of the phone: Phone works well inbound and outbound. Functions as intended. normal phone icon on the phone
• Thew second line, shows an icon with an X on the phone, outbound IS OK, inbound , nothing... No IP Address showing in endpoint manager
• the 3rd & 4th, work outbound, inbound straight to VM, phone Icon is Normal, but the IP address in endpoint manager is the ip address of the pbx server (hosted solution)...

So I'm thinking in the endpoint manager, the ipaddress should probably be that of the phone, since the only extention that is working properly is the one with the local ip address.

I'm a newbie, and not great with linux, hence the graphical interface... Any direction would be greatly appreciated.

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@expresspotato wrote:

Hello,

We have a queue setup on the latest FreePBX distro with all up to date modules on a few Yealink T46G phones. We setup a BLF to one of the function keys, with the extension as described here: wiki.freepbx.org/display/FPG/Queues+Module+User+Guide#QueuesModuleUserGuide-GeneralSettings

So the resulting BLF (with extension / value set to): *45*3000*100 for Extension 3000, Queue 100.

The guide describes 'Generate Device Hints' but there is no such option.

The BLF indicator does turn green, but doesn't change to any other color when extension 3000 joints the queue on the same phone or any other. Other BLF indicators such as busy extensions do work.

I do see the hint, but the state does not change to anything other than Unavailable.

Other people have mentioned enabling 'Enable Custom Device States' in the Settings -> Advanced, but its not there (found from here: http://community.freepbx.org/t/solved-blf-hints-not-working/13491/9)

Any ideas?

[root@freepbx ~]# asterisk -rx 'core show hints'

-= Registered Asterisk Dial Plan Hints =-

*8575@park-hints : park:75@parkedcalls State:Idle Presence:not_set Watchers 0
*8574@park-hints : park:74@parkedcalls State:Idle Presence:not_set Watchers 0
*8577@park-hints : park:77@parkedcalls State:Idle Presence:not_set Watchers 0
*8576@park-hints : park:76@parkedcalls State:Idle Presence:not_set Watchers 0
*8571@park-hints : park:71@parkedcalls State:Idle Presence:not_set Watchers 0
*8573@park-hints : park:73@parkedcalls State:Idle Presence:not_set Watchers 0
1001@ext-local : &Custom:DND1001,Cust State:Idle Presence:not_set Watchers 0
*8572@park-hints : park:72@parkedcalls State:Idle Presence:not_set Watchers 0
1002@ext-local : SIP/1002&Custom:DND1 State:Unavailable Presence:not_set Watchers 0
1003@ext-local : SIP/1003&Custom:DND1 State:Unavailable Presence:not_set Watchers 0
*8578@park-hints : park:78@parkedcalls State:Idle Presence:not_set Watchers 0
_45X.@ext-queues : ${DB(AMPUSER/${EXTEN State:Unavailable Presence: Watchers 0
_46X.@ext-queues : ${DB(AMPUSER/${EXTEN State:Unavailable Presence: Watchers 0
*271@timeconditions-: Custom:TC1 State:InUse Presence:not_set Watchers 0
885@ext-meetme : confbridge:885 State:Unavailable Presence:not_set Watchers 0
_*98X.@app-dialvm : MWI:${EXTEN:3}@${DB( State:Unavailable Presence: Watchers 0
72@park-hints : park:72@parkedcalls State:Idle Presence:not_set Watchers 0
73@park-hints : park:73@parkedcalls State:Idle Presence:not_set Watchers 0
70@park-hints : park:71@parkedcalls& State:Idle Presence:not_set Watchers 0
71@park-hints : park:71@parkedcalls State:Idle Presence:not_set Watchers 0
76@park-hints : park:76@parkedcalls State:Idle Presence:not_set Watchers 0
77@park-hints : park:77@parkedcalls State:Idle Presence:not_set Watchers 0
74@park-hints : park:74@parkedcalls State:Idle Presence:not_set Watchers 0
75@park-hints : park:75@parkedcalls State:Idle Presence:not_set Watchers 0
78@park-hints : park:78@parkedcalls State:Idle Presence:not_set Watchers 0
_*21X.@ext-findmefol: Custom:FOLLOWME${EXT State:Unavailable Presence: Watchers 0
46XXXX*200@ext-qu: Queue:${EXTEN:9}pau State:Unavailable Presence: Watchers 0
453000*100@ext-que: State:Unavailable Presence:not_set Watchers 1
_*80X.@ext-local : ${DB(AMPUSER/${EXTEN State:Unavailable Presence: Watchers 0
_*76X.@ext-dnd-hints: Custom:DEVDND${EXTEN State:Unavailable Presence: Watchers 0
_*96X.@ext-cf-hints : Custom:DEVCF${EXTEN: State:Unavailable Presence: Watchers 0
_45XXXXXXX@ext-que: Custom:QUEUE${EXTEN: State:Unavailable Presence: Watchers 0
3004@ext-local : SIP/3004&Custom:DND3 State:Idle Presence:not_set Watchers 0
3002@ext-local : SIP/3002&Custom:DND3 State:Idle Presence:not_set Watchers 1
3003@ext-local : SIP/3003&Custom:DND3 State:Idle Presence:not_set Watchers 1
3000@ext-local : SIP/3000&Custom:DND3 State:Idle Presence:not_set Watchers 2
46XXXX*100@ext-qu: Queue:${EXTEN:9}pau State:Unavailable Presence: Watchers 0
*87@ext-meetme : confbridge:885 State:Unavailable Presence:not_set Watchers 0

---

• 38 hints registered

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@mjurcev1 wrote:

Tried to update from System Admin, the asterisk crashed. Restarted the whole system.
After that I am trying to update from command line with "upgrade-10.13.66-17.sh" script.
It hangs on the following part:
Downloading & Installing ucpnode...
Starting ucpnode download..
Processing ucpnode
Verifying local module download...Verified
Extracting...Done
Module ucpnode successfully downloaded
Installing/Updating Required Libraries. This may take a while.....

Obviously, nothing happens for a couple of hours...tried to restart the system, but the same thing happened again. Since this never happened before, please advise further actions.

Here is the whole screen captured:

This appears to be a FreePBX Distro system as it has a Distro Version of 10.13.66-16

Your FreePBX Distro System is being upgraded to 10.13.66-17. Please standby...

STAGE 1 STARTING - GUI Modules

Upgrade All FreePBX GUI Modules

trying to run as user asterisk:

[AMPSBIN] already set to [/usr/sbin]

Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown +Conf
Setting Permissions...
Collecting Files...Done
48328/48328 [============================] 100%
Finished setting permissions
The following error(s) occured:
- Module not installed: cannot uninstall
Module restart successfully deleted
Updating Hooks...Done
No repos specified, using: [standard,unsupported,extended,commercial] from last GUI settings

Up to date.
Installing: restart, ucpnode
Downloading & Installing restart...
Starting restart download..
Processing restart
Verifying local module download...Verified
Extracting...Done
Module restart successfully downloaded
Unable to install module restart:
- Failed to install Bulk Phone Restart due to the following conflicting module(s): EndPoint Manager
Updating Hooks...Done

Downloading & Installing ucpnode...
Starting ucpnode download..
Processing ucpnode
Verifying local module download...Verified
Extracting...Done
Module ucpnode successfully downloaded
Installing/Updating Required Libraries. This may take a while.....

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@cekpek wrote:

Hey Guys,

I am trying to setup an incoming route which come from group of trunk to the IVR.

First, i have an OpenVox 8 FXO PORTS which i have set it up and configured the group in dahdi-channels.conf which 6 ports grouped into group 0 and another 2 ports grouped to group 1.

Group 0 was intended for the land line purpose and the other group will be use with the VOIP Gateway.

So i am trying to achieve which any call that hit the trunk in Group 0 will be transfered to IVR.

Anyone can give the idea how to do that?

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@expresspotato wrote:

Hi All,

We're trying to enable HD Voice (G.722) here between office phones have have selected G.722 in the codecs list found in the Asterisk SIP settings page. This does work from office phone to office phone, but the trunk only supports G.711.

Thought it would just work, as our voip provider settings we have just G.711u checked.

No such luck. So we tried adding the following to the trunk config in sip settings -> outgoing:
disallow=all
allow=ulaw

The outside party can hear us, but we cannot hear them.

Any ideas?

`[root@localhost ~]# cat /var/log/asterisk/full

[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [14168221123@from-internal:1] Macro("PJSIP/3000-00000018", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/3000-00000018", "TOUCH_MONITOR=1482826681.59") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/3000-00000018", "AMPUSER=3000") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/3000-00000018", "0?report") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/3000-00000018", "1?Set(REALCALLERIDNUM=3000)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/3000-00000018", "AMPUSER=3000") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/3000-00000018", "0?limit") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/3000-00000018", "AMPUSERCIDNAME=Kevin") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("PJSIP/3000-00000018", "0?report") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:9] Set("PJSIP/3000-00000018", "AMPUSERCID=3000") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:10] Set("PJSIP/3000-00000018", "_DIALOPTIONS=Ttr") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:11] Set("PJSIP/3000-00000018", "CALLERID(all)="Kevin" <3000>") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:12] GotoIf("PJSIP/3000-00000018", "0?limit") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:13] ExecIf("PJSIP/3000-00000018", "1?Set(GROUP(concurrency_limit)=3000)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:14] ExecIf("PJSIP/3000-00000018", "0?Set(CHANNEL(language)=)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:15] GotoIf("PJSIP/3000-00000018", "1?continue") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (macro-user-callerid,s,29)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:29] Set("PJSIP/3000-00000018", "CALLERID(number)=3000") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:30] Set("PJSIP/3000-00000018", "CALLERID(name)=Kevin") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:31] GotoIf("PJSIP/3000-00000018", "0?cnum") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:32] Set("PJSIP/3000-00000018", "CDR(cnam)=Kevin") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:33] Set("PJSIP/3000-00000018", "CDR(cnum)=3000") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-user-callerid:34] Set("PJSIP/3000-00000018", "CHANNEL(language)=en") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [14168221123@from-internal:2] Gosub("PJSIP/3000-00000018", "sub-record-check,s,1(out,14168221123,dontcare)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/3000-00000018", "0?initialized") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:2] Set("PJSIP/3000-00000018", "_RECSTATUS=INITIALIZED") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:3] Set("PJSIP/3000-00000018", "NOW=1482826681") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:4] Set("PJSIP/3000-00000018", "__DAY=27") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:5] Set("PJSIP/3000-00000018", "__MONTH=12") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:6] Set("PJSIP/3000-00000018", "__YEAR=2016") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:7] Set("PJSIP/3000-00000018", "__TIMESTR=20161227-031801") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:8] Set("PJSIP/3000-00000018", "__FROMEXTEN=3000") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:9] Set("PJSIP/3000-00000018", "_MONFMT=wav") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/3000-00000018", "Recordings initialized") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/3000-00000018", "0?Set(ARG3=dontcare)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/3000-00000018", "REC_POLICY_MODE_SAVE=") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/3000-00000018", "0?Set(REC_STATUS=NO)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/3000-00000018", "3?checkaction") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (sub-record-check,s,17)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/3000-00000018", "1?sub-record-check,out,1") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (sub-record-check,out,1)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [out@sub-record-check:1] NoOp("PJSIP/3000-00000018", "Outbound Recording Check from 3000 to 14168221123") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [out@sub-record-check:2] Set("PJSIP/3000-00000018", "RECMODE=dontcare") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [out@sub-record-check:3] ExecIf("PJSIP/3000-00000018", "1?Goto(routewins)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (sub-record-check,out,7)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [out@sub-record-check:7] Gosub("PJSIP/3000-00000018", "recordcheck,1(dontcare,out,14168221123)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/3000-00000018", "Starting recording check against dontcare") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/3000-00000018", "dontcare") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [recordcheck@sub-record-check:3] Return("PJSIP/3000-00000018", "") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [out@sub-record-check:8] Return("PJSIP/3000-00000018", "") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [14168221123@from-internal:3] ExecIf("PJSIP/3000-00000018", "0 ?Set(CDR(accountcode)=)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [14168221123@from-internal:4] Set("PJSIP/3000-00000018", "MOHCLASS=default") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [14168221123@from-internal:5] ExecIf("PJSIP/3000-00000018", "0?Set(TRUNKCIDOVERRIDE=Helium Investments <18778435486>)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [14168221123@from-internal:6] Set("PJSIP/3000-00000018", "_NODEST=") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [14168221123@from-internal:7] Macro("PJSIP/3000-00000018", "dialout-trunk,2,14168221123,,off") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:1] Set("PJSIP/3000-00000018", "DIAL_TRUNK=2") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:2] GosubIf("PJSIP/3000-00000018", "0?sub-pincheck,s,1()") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:3] GotoIf("PJSIP/3000-00000018", "0?disabletrunk,1") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:4] Set("PJSIP/3000-00000018", "DIAL_NUMBER=14168221123") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:5] Set("PJSIP/3000-00000018", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:6] Set("PJSIP/3000-00000018", "OUTBOUND_GROUP=OUT_2") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:7] GotoIf("PJSIP/3000-00000018", "1?nomax") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (macro-dialout-trunk,s,9)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf("PJSIP/3000-00000018", "0?skipoutcid") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:10] Set("PJSIP/3000-00000018", "DIAL_TRUNK_OPTIONS=T") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:11] Macro("PJSIP/3000-00000018", "outbound-callerid,2") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:1] ExecIf("PJSIP/3000-00000018", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:2] ExecIf("PJSIP/3000-00000018", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:3] ExecIf("PJSIP/3000-00000018", "0?Set(REALCALLERIDNUM=3000)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:4] GotoIf("PJSIP/3000-00000018", "1?normcid") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (macro-outbound-callerid,s,7)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:7] Set("PJSIP/3000-00000018", "USEROUTCID=Helium Investments <18778435486>") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:8] Set("PJSIP/3000-00000018", "EMERGENCYCID=") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:9] Set("PJSIP/3000-00000018", "TRUNKOUTCID=<18778435486>") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:10] GotoIf("PJSIP/3000-00000018", "1?trunkcid") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (macro-outbound-callerid,s,15)
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:15] ExecIf("PJSIP/3000-00000018", "1?Set(CALLERID(all)=<18778435486>)") in new stack
[2016-12-27 03:18:01] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:16] ExecIf("PJSIP/3000-00000018", "1?Set(CALLERID(all)=Helium Investments <18778435486>)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:17] ExecIf("PJSIP/3000-00000018", "0?Set(CALLERID(all)=)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:18] ExecIf("PJSIP/3000-00000018", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:19] ExecIf("PJSIP/3000-00000018", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:20] Set("PJSIP/3000-00000018", "CDR(outbound_cnum)=18778435486") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-outbound-callerid:21] Set("PJSIP/3000-00000018", "CDR(outbound_cnam)=Helium Investments") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:12] GosubIf("PJSIP/3000-00000018", "0?sub-flp-2,s,1()") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:13] Set("PJSIP/3000-00000018", "OUTNUM=14168221123") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:14] Set("PJSIP/3000-00000018", "custom=SIP/P2659570569") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:15] ExecIf("PJSIP/3000-00000018", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf("PJSIP/3000-00000018", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:17] Macro("PJSIP/3000-00000018", "dialout-trunk-predial-hook,") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/3000-00000018", "") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/3000-00000018", "0?bypass,1") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/3000-00000018", "1?Set(CONNECTEDLINE(num,i)=14168221123)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/3000-00000018", "1?Set(CONNECTEDLINE(name,i)=CID:18778435486)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/3000-00000018", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)18778435486)") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/3000-00000018", "0?customtrunk") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-dialout-trunk:23] Dial("PJSIP/3000-00000018", "SIP/P2659570569/14168221123,300,T") in new stack
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] netsock2.c: Using SIP RTP TOS bits 184
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] netsock2.c: Using SIP RTP CoS mark 5
[2016-12-27 03:18:02] VERBOSE[14556][C-00000011] app_dial.c: Called SIP/P2659570569/14168221123
[2016-12-27 03:18:03] VERBOSE[14556][C-00000011] app_dial.c: SIP/P2659570569-00000009 is making progress passing it to PJSIP/3000-00000018
[2016-12-27 03:18:03] VERBOSE[14556][C-00000011] app_dial.c: SIP/P2659570569-00000009 answered PJSIP/3000-00000018
[2016-12-27 03:18:03] VERBOSE[14568][C-00000011] bridge_channel.c: Channel SIP/P2659570569-00000009 joined 'simple_bridge' basic-bridge <0dfaf728-69ec-4393-b084-13abcfe8f993>
[2016-12-27 03:18:03] VERBOSE[14556][C-00000011] bridge_channel.c: Channel PJSIP/3000-00000018 joined 'simple_bridge' basic-bridge <0dfaf728-69ec-4393-b084-13abcfe8f993>
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] bridge_channel.c: Channel PJSIP/3000-00000018 left 'simple_bridge' basic-bridge <0dfaf728-69ec-4393-b084-13abcfe8f993>
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'PJSIP/3000-00000018' in macro 'dialout-trunk'
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] pbx.c: Spawn extension (from-internal, 14168221123, 7) exited non-zero on 'PJSIP/3000-00000018'
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] pbx.c: Executing [h@from-internal:1] Macro("PJSIP/3000-00000018", "hangupcall") in new stack
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/3000-00000018", "1?theend") in new stack
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2016-12-27 03:18:10] VERBOSE[14568][C-00000011] bridge_channel.c: Channel SIP/P2659570569-00000009 left 'simple_bridge' basic-bridge <0dfaf728-69ec-4393-b084-13abcfe8f993>
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("PJSIP/3000-00000018", "0?Set(CDR(recordingfile)=)") in new stack
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] pbx.c: Executing [s@macro-hangupcall:4] Hangup("PJSIP/3000-00000018", "") in new stack
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/3000-00000018' in macro 'hangupcall'
[2016-12-27 03:18:10] VERBOSE[14556][C-00000011] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/3000-00000018'
`

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@many_miles wrote:

Background
Yesterday, I transferred my PBX from one hardware to another and upgraded it.

On the old box, I was using:
FreePBX 12.0.76.4
6.12.65-32

On the replacement box, I installed 6.12.65-30, patched it up to -32 and then did a restore from a backup of the above box.

I then upgraded the replacement box using the upgradescripts

The replacement box is now at
FreePBX 13.0.190.7
10.13.66-17

I now have a problem where "fwconsole restart" borks up my BLFs. If I do an fwconsole r immediately after the fwconsole restart, the BLFs start working again.

You might suggest, "Don't do an 'fwconsole restart'". I can avoid that, but, when I reboot the machine, I am in the same borked up BLF situation and have to run an fwconsole r to restore proper operation. Again, not a big deal if I'm around, but, if we have a power failure and I'm not around, BLFs will remained broken.

Steps to reproduce:
fwconsole restart
asterisk -rx 'core show hints' | grep ext-local | wc -l
That produces a count of "4".

Resolution steps
fwconsole r
asterisk -rx 'core show hints' | grep ext-local | wc -l
That now produces the expected count of 133.

Any suggestions?

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@robdgem1 wrote:

Hi All

I have setup a new FreePBX system which is working well except when dialing numbers from our address book.
There are two issues that I am experiencing.
The first is that the numbers are stored with the spaces between the numbers, ie 089 123 1234, when calling this number the calls fail.
Secondly, some numbers have the full international number +27 89 123 1234, which also fails.

Does anyone have some basic instructions for a FreePBX noob to firstly remove the spaces and secondly change the +27 to a 0?

Any help would be greatly appreciated.

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@AMWild wrote:

I'm trying to install Freepbx with Asterisk 14 into a docker container based on CentOS 6, running on a Synology RS812+ box with DSM 6 and a pre-existing mariadb. I've got DAHDI and all the other prerequisites compiling and installing.
I install Asterisk and:

chown -R $ASTERISKUSER. -v /var/run/asterisk
chown -R $ASTERISKUSER. -v /etc/asterisk
chown -R $ASTERISKUSER. -v /var/{lib,log,spool}/asterisk
chown -R $ASTERISKUSER. -v /usr/lib64/asterisk
chown -R $ASTERISKUSER. -v /var/www/
chown -R $ASTERISKUSER. -v /var/www/*

However, when I get to FreePBX:

./install -n -v --dbuser=asterisk

I get the error:

Error!
Could not determine Asterisk version (got: Unable to access the running directory (Permission denied). Changing to '/' for compatibility.). Please report this.

And then:

...
Finished setting permissions
Generating default configurations...
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 11
Stack trace:
1. Exception->() /var/lib/asterisk/bin/retrieve_conf:11

And finally, I:

fwconsole restart

And get:

Running FreePBX shutdown...

[Whoops\Exception\ErrorException]
fclose(): 61 is not a valid stream resource

restart [-i|--immediate] [args1] ... [argsN]

SELinux is disabled, and iptables is set to allow all incoming and outgoing connections by default and has no other settings assigned.

The docker container is running as --privileged and
How can I overcome this problem?

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@expresspotato wrote:

Hi all,

We're running the latest FreePBX Distro with all modules up to date and decided to enable voicemail on the extensions. Unfortunately nothing is played and there is only silence on the line. DND or timeout, same result.

The individual extension has not setup their voicemail, but I expected some default to be played...

Any ideas?

I don't even see it Playing anything like some other tickets I've found on here...:

See full log on patebin: http://pastebin.com/VJufvn5Q

Edit: There is no audio either when dailing *97? I

Solved: Having the G.722 codec enabled caused no audio. It is the first on the list from the phone, but the 3rd on the list in Asterisk SIP Settings and enabled (now not even checked). Not sure why there was no audio still, as I'd like to move to G.722 internally anyways.

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@Cricchetto wrote:

Hello everyone, I look for the solution to a problem. PBX 12, when a call to voicemail, part a voice that says: "the called party is busy on another call, leave a message after the beep." I need to change that message or delete it so that you only hear the beep. You have any suggestions or ideas?
Thank you

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@davidbqzt wrote:

Hi everybody,

In an outbound route, you have 'Route Password' and 'Pin Set'. Into the 'Route Password' help says: "A numerical password, or the path to an Authenticate password file can be used."
I want to know how to use specify the passwod file, because I create a password file, for instance: '/etc/asterisk/pinset_codigoPersonal', but when I write down the path on 'Route password' and try to submit it, I received an error that says: 'Route password must be numeric or leave blank to disable'.

I know that I can get similar behavior using Pin Set, but I want to use Route Password, acctually what I expect is that two passwords are required before call, one using 'Route password' and the other by using 'Pin Set'.

Thanks for your help.

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@michelpy wrote:

I Was just reading this :

I re-created it;

in extensions_custom.conf
[automap]
include => from-did-direct

Then I created a custom destination named automap in FreePBX:
automap,${CALLERID(DNID):-4},1
In my case, the extension is the last 4 digits of the DID.

Finally I used the custom destination in the inbound route in FreePBX.

It works ! Only one inbound route !

Now I have a dumb question : how do I modify this so it does the following :
- If the extension exists, send it to the extension.
- If the extension does not exist, send it to a trunk.

Please enlighten me.

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@FogWatch wrote:

When I choose to display my extensions, FreePBX 13 returns a stack of errors in red boxes on the right hand side that include "Undefined index: data".

In the JSON panel on the right hand side of my Firefox 45.6.0 Web Console the error is displayed as:

error: Object
type: "Whoops\Exception\ErrorException"
message: "Undefined index: data"
file: "/var/www/html/admin/libraries/php-asmanager.php"
line: 1525

Why can't I display my extensions?

---

FreePBX 13.0.190.7 runs in a privileged LXC container. Both host and container run Gentoo.

uname -r
4.4.26-gentoo

lxc-info --version
2.0.6

php -v
PHP 5.6.28-pl0-gentoo (cli) (built: Dec 24 2016 22:44:11)
Copyright (c) 1997-2016 The PHP Group
Zend Engine v2.6.0, Copyright (c) 1998-2016 Zend Technologies
with Zend OPcache v7.0.6-dev, Copyright (c) 1999-2016, by Zend Technologies

mysql -V
mysql Ver 15.1 Distrib 10.0.28-MariaDB, for Linux (x86_64) using readline 5.1

apache2 -V
Server version: Apache/2.4.25 (Unix)
Server built: Dec 24 2016 22:23:25
Server's Module Magic Number: 20120211:67
Server loaded: APR 1.5.2, APR-UTIL 1.5.4
Compiled using: APR 1.5.2, APR-UTIL 1.5.4
Architecture: 64-bit
Server MPM: prefork
threaded: no
forked: yes (variable process count)
Server compiled with....
-D APR_HAS_SENDFILE
-D APR_HAS_MMAP
-D APR_HAVE_IPV6 (IPv4-mapped addresses enabled)
-D APR_USE_SYSVSEM_SERIALIZE
-D APR_USE_PTHREAD_SERIALIZE
-D SINGLE_LISTEN_UNSERIALIZED_ACCEPT
-D APR_HAS_OTHER_CHILD
-D AP_HAVE_RELIABLE_PIPED_LOGS
-D DYNAMIC_MODULE_LIMIT=256
-D HTTPD_ROOT="/usr"
-D SUEXEC_BIN="/usr/bin/suexec"
-D DEFAULT_PIDLOG="/var/run/httpd.pid"
-D DEFAULT_SCOREBOARD="logs/apache_runtime_status"
-D DEFAULT_ERRORLOG="logs/error_log"
-D AP_TYPES_CONFIG_FILE="/etc/apache2/mime.types"
-D SERVER_CONFIG_FILE="/etc/apache2/httpd.conf"

asterisk -V
Asterisk 13.13.1

Roughly followed the Ubuntu guildeline.

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@vettalex wrote:

Hi, I just joined this forum,
I have a Problemas with the FreePBX Program Activation, can you help me please?
I installed the 10/13/66 version of FreePBX on a dual-core PC with 1 GB of RAM, then, once created the extensions that I need to activate the program, I went to the "Admin" - "System Admin" and I clicked on in the upper right button "Activation"; then you see the activation page, which tells me that the machine is not turned on and that you have to press on the "Active" button to activate it, but when I click, appearance that the page loads, and I get the same page again without to change something. I can not activate the program, how can I do? Thank you all for the help

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@sco01 wrote:

SIP trunk. No outbound calls

Hi,

I'm having a problem that is driving me nuts and my Google-fu is apparently limited. I have FreePBX 13.0.190.7 (The Distro) with Asterisk 13.13.1 and a bunch of Cisco SIP (chan-sip) and SCCP phones (and a Zoiper soft-phone configured using chan-pjsip). The pbx and all phones are sitting behind NAT and a dynamic IP adress. I have a SIP trunk from a local provider. I can make calls between all phones internally

FreePBX is registered with the provider and I can receive external calls to the DID. Audio is working correctly for all calls both internally and from the trunk. The problem is that I can't call any external numbers. I have tried to narrow the problem down and a SIP debug shows that i get a “SIP/2.0 403 Forbidden” from the voip provider whenever I try to call out over the trunk. I initially used chan_pjsip for the trunk but switched over to chan_sip but the result is the same.

Here are the PEER Details on the outgoing tab of the trunk sip settings

host=95.143.207.218
port=9950
username=3976
secret=hunter2
type=peer
qualify=yes
nat=yes
insecure=port,invite

I realise that there must be something wrong with my configuration but I haven't been able to nail it yet so I would really appreciate any help I can get with this.

Here is a sip debug trace from a connection attempt. What else can I provide to narrow down the issue?

SIP Debugging enabled
[2017-01-02 12:23:39] WARNING[7183]: res_pjsip_pubsub.c:3150 pubsub_on_rx_publish_request: No registered publish handler for event presence
[2017-01-02 12:23:39] WARNING[7183]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
[2017-01-02 12:23:39] WARNING[2055]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
Audio is at 12034
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 95.143.207.218:9950:
INVITE sip:033287301@95.143.207.218:9950 SIP/2.0
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK322b0c61;rport
Max-Forwards: 70
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
To: <sip:033287301@95.143.207.218:9950>
Contact: <sip:3976@83.233.167.184:5160>
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 102 INVITE
User-Agent: FPBX-13.0.190.7(13.13.1)
Date: Mon, 02 Jan 2017 11:23:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 1978262473 1978262473 IN IP4 83.233.167.184
s=Asterisk PBX 13.13.1
c=IN IP4 83.233.167.184
t=0 0
m=audio 12034 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
Retransmitting #1 (NAT) to 95.143.207.218:9950:
INVITE sip:033287301@95.143.207.218:9950 SIP/2.0
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK322b0c61;rport
Max-Forwards: 70
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
To: <sip:033287301@95.143.207.218:9950>
Contact: <sip:3976@83.233.167.184:5160>
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 102 INVITE
User-Agent: FPBX-13.0.190.7(13.13.1)
Date: Mon, 02 Jan 2017 11:23:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 1978262473 1978262473 IN IP4 83.233.167.184
s=Asterisk PBX 13.13.1
c=IN IP4 83.233.167.184
t=0 0
m=audio 12034 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:95.143.207.218:9950 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK322b0c61;rport=5160
To: <sip:033287301@95.143.207.218:9950>
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:95.143.207.218:9950 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK322b0c61;rport=5160
Proxy-Authenticate: Digest nonce="414d535c0e7ac93323:88273b0cbea406e86e0ccb0dc2de0a05",algorithm=MD5,realm="3CXPhoneSystem"
To: <sip:033287301@95.143.207.218:9950>;tag=bc720b74
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem 15.0.59950.0 (59299)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 95.143.207.218:9950:
ACK sip:033287301@95.143.207.218:9950 SIP/2.0
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK322b0c61;rport
Max-Forwards: 70
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
To: <sip:033287301@95.143.207.218:9950>;tag=bc720b74
Contact: <sip:3976@83.233.167.184:5160>
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 102 ACK
User-Agent: FPBX-13.0.190.7(13.13.1)
Content-Length: 0


---
Audio is at 12034
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 95.143.207.218:9950:
INVITE sip:033287301@95.143.207.218:9950 SIP/2.0
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK3cc9854d;rport
Max-Forwards: 70
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
To: <sip:033287301@95.143.207.218:9950>
Contact: <sip:3976@83.233.167.184:5160>
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 103 INVITE
User-Agent: FPBX-13.0.190.7(13.13.1)
Proxy-Authorization: Digest username="3976", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:033287301@95.143.207.218:9950", nonce="414d535c0e7ac93323:88273b0cbea406e86e0ccb0dc2de0a05", response="a027db0748189532f8ac2dd211a3c44f"
Date: Mon, 02 Jan 2017 11:23:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 1978262473 1978262474 IN IP4 83.233.167.184
s=Asterisk PBX 13.13.1
c=IN IP4 83.233.167.184
t=0 0
m=audio 12034 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:95.143.207.218:9950 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK322b0c61;rport=5160
Proxy-Authenticate: Digest nonce="414d535c0e7ac93323:88273b0cbea406e86e0ccb0dc2de0a05",algorithm=MD5,realm="3CXPhoneSystem"
To: <sip:033287301@95.143.207.218:9950>;tag=bc720b74
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem 15.0.59950.0 (59299)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 95.143.207.218:9950:
ACK sip:033287301@95.143.207.218:9950 SIP/2.0
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK3cc9854d;rport
Max-Forwards: 70
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
To: <sip:033287301@95.143.207.218:9950>
Contact: <sip:3976@83.233.167.184:5160>
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 102 ACK
User-Agent: FPBX-13.0.190.7(13.13.1)
Content-Length: 0


---

<--- SIP read from UDP:95.143.207.218:9950 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK3cc9854d;rport=5160
To: <sip:033287301@95.143.207.218:9950>;tag=237b1917
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 103 INVITE
User-Agent: 3CXPhoneSystem 15.0.59950.0 (59299)
Warning: 499 WIN-F6MOI2921OH "Caller is forbidden"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 95.143.207.218:9950:
ACK sip:033287301@95.143.207.218:9950 SIP/2.0
Via: SIP/2.0/UDP 83.233.167.184:5160;branch=z9hG4bK3cc9854d;rport
Max-Forwards: 70
From: <sip:3976@95.143.207.218:5160>;tag=as29ca084e
To: <sip:033287301@95.143.207.218:9950>;tag=237b1917
Contact: <sip:3976@83.233.167.184:5160>
Call-ID: 2a09d8733596b541708a8e2e336c4d39@95.143.207.218
CSeq: 103 ACK
User-Agent: FPBX-13.0.190.7(13.13.1)
Content-Length: 0


---
[2017-01-02 12:23:39] WARNING[2405][C-00000003]: chan_sip.c:23861 handle_response_invite: Received response: "Forbidden" from '<sip:3976@95.143.207.218:5160>;tag=as29ca084e'
Scheduling destruction of SIP dialog '2a09d8733596b541708a8e2e336c4d39@95.143.207.218' in 7616 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.120:5160:
OPTIONS sip:3@192.168.0.120:5160;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.38:5160;branch=z9hG4bK093cf21c
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.38:5160>;tag=as66e16942
To: <sip:3@192.168.0.120:5160;transport=udp>
Contact: <sip:Unknown@192.168.0.38:5160>
Call-ID: 31bb6bb22132d9681c332f32069407d9@192.168.0.38:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.13.1)
Date: Mon, 02 Jan 2017 11:23:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.120:51837 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.38:5160;branch=z9hG4bK093cf21c
From: "Unknown" <sip:Unknown@192.168.0.38:5160>;tag=as66e16942
To: <sip:3@192.168.0.120:5160;transport=udp>;tag=0018187f3bf514043c773463-6a31ccd8
Call-ID: 31bb6bb22132d9681c332f32069407d9@192.168.0.38:5160
Date: Mon, 02 Jan 2017 11:23:39 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7941G/9.4.2
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0
Content-Length: 265
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 6727 0 IN IP4 192.168.0.120
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 102 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 12 lines) ---

Posts: 12

Participants: 4

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@esteven wrote:

Hello;

This is my first post and I would want to explain myself clearly, since I'm new to FreePBX. I started to use it about 3 months ago and I learned lots of things but I stuck with one concrete issue I don't seem to manage to solve because I could not find an answer looking all the day through Google. Here I go:

I have 5 trunks, each one has a telephone number. Let's say Trunk1 has 1111, Trunk2 has 2222 and so on.

I have 100 SIP extensions. Let's say from 1 to 100.

I have 5 Outbound Routes, each one going to each of the five Trunks. Let's say that the patterns to use each one are prefixes 1,2,3,4 and 5, so for example when I want to call some landline or cellphone through the number 1111 I have to dial 1 + country code + number.

What I want to accomplish is to make some specific extensions don't need to dial the Outbound Route pattern in order to make calls. For example, make extensions 1 to 10 call directly dialing with country code + number through the Outbound Route I configure, like an autocomplete.

Why this? because we have 5 departments and I want each department use the corresponding trunk without needing to dial any pattern, just by configuring which extensions will use which outbound route.

I hope I made myself clear.

Thank you very much for your help!

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Participants: 2

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@cekpek wrote:

I have FreePBX box setup without any problem, everything is working except for the call pickup.

I tried to use the code *8 followed by the extension, or ** followed by extension seems not to work at all.

I am using Linksys ATA Adapter PAP2T which i have emptied the codec in the linksys box itself.

Anyone else having the same issue?

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Participants: 1

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