Paging groutps don't hang up

We’re testing a new install with FreePBX 2.10 RC and found when using paging groups the extensions don’t hang up after the page is complete. Has anybody seen this behavior?

I’ll post logs in a few hours when I get to the site again.

Need more info. Be specific about what versions of FreePBX and Asterisk you are using. Is this a distro install or built by hand?

What type of paging? Are you paging via the phones? Are you using an ATA to integrate with overhead paging?

Sorry for the lack of details. We’re using piaf 1.7.5.6 with Asterisk 1.6.2.20.

We’re using Polycom phones, no ATAs at the moment but will add once this is working.

I’ve tested paging with individual phones and it works fine. The problem is only when using a paging group. I created a paging group, 600 and when I dial 600 from any extension all other extensions pick correctly. I’ve tried both with duplex and without with the same behavior.

Here’s a part of the log calling paging group 600 from extension 105.

Info: Ring Answer") in new stack
– Executing [[email protected]:3] Set(“SIP/105-00000127”, “_CALLINFO=Call-Info: ;answer-after=0”) in new stack
– Executing [[email protected]:4] Set(“SIP/105-00000127”, “_SIPURI=intercom=true”) in new stack
– Executing [[email protected]:5] Set(“SIP/105-00000127”, “_DOPTIONS=A(beep)”) in new stack
– Executing [[email protected]:6] Set(“SIP/105-00000127”, “_DTIME=5”) in new stack
– Executing [[email protected]:7] Set(“SIP/105-00000127”, “_ANSWERMACRO=”) in new stack
– Executing [[email protected]:8] Set(“SIP/105-00000127”, “PAGE_CONF=1327607975480”) in new stack
– Executing [[email protected]:9] Return(“SIP/105-00000127”, “”) in new stack
– Executing [[email protected]:7] Set(“SIP/105-00000127”, “PAGEMODE=PAGE”) in new stack
– Executing [[email protected]:8] Set(“SIP/105-00000127”, “PAGE_MEMBERS=102-103-104-105-201-202-301-302-303”) in new stack
– Executing [[email protected]:9] Set(“SIP/105-00000127”, “PAGE_CONF_OPTS=1qsm”) in new stack
– Executing [[email protected]:10] AGI(“SIP/105-00000127”, “page.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/page.agi

-- Executing [[email protected]:1] Macro("Local/[email protected];2", "autoanswer,102") in new stack
-- Executing [[email protected]:1] Macro("Local/[email protected];2", "autoanswer,103") in new stack
-- Executing [[email protected]:1] Set("Local/[email protected];2", "DIAL=SIP/102") in new stack
-- Executing [[email protected]:2] ExecIf("Local/[email protected];2", "0?Set(DIAL=DAHDI/102)") in new stack
-- Executing [[email protected]:1] Set("Local/[email protected];2", "DIAL=SIP/103") in new stack
-- Executing [[email protected]:2] ExecIf("Local/[email protected];2", "0?Set(DIAL=DAHDI/103)") in new stack
-- Executing [[email protected]:3] GotoIf("Local/[email protected];2", "0?macro") in new stack
-- Executing [[email protected]:1] Macro("Local/[email protected];2", "autoanswer,104") in new stack
-- Executing [[email protected]:3] GotoIf("Local/[email protected];2", "0?macro") in new stack
-- Executing [[email protected]:4] Set("Local/[email protected];2", "phone=PolycomSoundPointIP-SPIP_550-UA/3.3.2.0413") in new stack
-- Executing [[email protected]:5] ExecIf("Local/[email protected];2", "0?Set(CALLINFO=Call-Info: <sip:broadworks.net>\;answer-after=0)") in new stack
-- Executing [[email protected]:6] ExecIf("Local/[email protected];2", "0?Set(ALERTINFO=Alert-Info: Intercom)") in new stack
-- Executing [[email protected]:7] ExecIf("Local/[email protected];2", "1?SipAddHeader(Alert-Info: Ring Answer)") in new stack
-- Executing [[email protected]:4] Set("Local/[email protected];2", "phone=PolycomSoundPointIP-SPIP_550-UA/3.3.2.0413") in new stack
-- Executing [[email protected]:5] ExecIf("Local/[email protected];2", "0?Set(CALLINFO=Call-Info: <sip:broadworks.net>\;answer-after=0)") in new stack
-- Executing [[email protected]:6] ExecIf("Local/[email protected];2", "0?Set(ALERTINFO=Alert-Info: Intercom)") in new stack
-- Executing [[email protected]:7] ExecIf("Local/[email protected];2", "1?SipAddHeader(Alert-Info: Ring Answer)") in new stack
-- Executing [[email protected]:1] Set("Local/[email protected];2", "DIAL=SIP/104") in new stack
-- Executing [[email protected]:8] ExecIf("Local/[email protected];2", "1?SipAddHeader(Call-Info: <uri>;answer-after=0)") in new stack
-- Executing [[email protected]:8] ExecIf("Local/[email protected];2", "1?SipAddHeader(Call-Info: <uri>;answer-after=0)") in new stack
-- Executing [[email protected]:1] Macro("Local/[email protected];2", "autoanswer,301") in new stack
-- Executing [[email protected]:1] Set("Local/[email protected];2", "DIAL=SIP/301") in new stack
-- Executing [[email protected]:9] ExecIf("Local/[email protected];2", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack
-- Executing [[email protected]:2] ExecIf("Local/[email protected];2", "0?Set(DIAL=DAHDI/104)") in new stack
-- Executing [[email protected]:3] GotoIf("Local/[email protected];2", "0?macro") in new stack
-- Executing [[email protected]:2] ExecIf("Local/[email protected];2", "0?Set(DIAL=DAHDI/301)") in new stack
-- Executing [[email protected]:4] Set("Local/[email protected];2", "phone=PolycomSoundPointIP-SPIP_550-UA/3.3.2.0413") in new stack
-- Executing [[email protected]:3] GotoIf("Local/[email protected];2", "0?macro") in new stack
-- Executing [[email protected]:4] Set("Local/[email protected];2", "phone=PolycomSoundPointIP-SPIP_650-UA/3.3.2.0413") in new stack
-- Executing [[email protected]:5] ExecIf("Local/[email protected];2", "0?Set(CALLINFO=Call-Info: <sip:broadworks.net>\;answer-after=0)") in new stack
-- Executing [[email protected]:6] ExecIf("Local/[email protected];2", "0?Set(ALERTINFO=Alert-Info: Intercom)") in new stack
-- Executing [[email protected]:7] ExecIf("Local/[email protected];2", "1?SipAddHeader(Alert-Info: Ring Answer)") in new stack
-- Executing [[email protected]:8] ExecIf("Local/[email protected];2", "1?SipAddHeader(Call-Info: <uri>;answer-after=0)") in new stack
-- Executing [[email protected]:9] ExecIf("Local/[email protected];2", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack
-- Executing [[email protected]:1] Macro("Local/[email protected];2", "autoanswer,303") in new stack

-- Executing [[email protected]:9] ExecIf("Local/[email protected];2", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack
-- Executing [[email protected]:2] Dial("Local/[email protected];2", "SIP/301,5,A(beep)") in new stack
-- Executing [[email protected]:2] Dial("Local/[email protected];2", "SIP/102,5,A(beep)") in new stack
-- Executing [[email protected]:5] ExecIf("Local/[email protected];2", "0?Set(CALLINFO=Call-Info: <sip:broadworks.net>\;answer-after=0)") in new stack

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Wait(“Local/[email protected];2”, “1”) in new stack
– Executing [[email protected]:2] Dial(“Local/[email protected];2”, “SIP/103,5,A(beep)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called 301
– Called 102
– Executing [[email protected]:6] ExecIf(“Local/[email protected];2”, “0?Set(ALERTINFO=Alert-Info: Intercom)”) in new stack
– Executing [[email protected]:7] ExecIf(“Local/[email protected];2”, “1?SipAddHeader(Alert-Info: Ring Answer)”) in new stack
== Using SIP RTP TOS bits 184
– Executing [[email protected]:8] ExecIf(“Local/[email protected];2”, “1?SipAddHeader(Call-Info: ;answer-after=0)”) in new stack
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Set(“Local/[email protected];2”, “DIAL=SIP/303”) in new stack
– Executing [[email protected]:9] ExecIf(“Local/[email protected];2”, “1?Set(__SIP_URI_OPTIONS=intercom=true)”) in new stack
– Executing [[email protected]:2] ExecIf(“Local/[email protected];2”, “0?Set(DIAL=DAHDI/303)”) in new stack
– Called 103
– Executing [[email protected]:3] GotoIf(“Local/[email protected];2”, “0?macro”) in new stack
– Executing [[email protected]:2] Dial(“Local/[email protected];2”, “SIP/104,5,A(beep)”) in new stack
– Executing [[email protected]:4] Set(“Local/[email protected];2”, “phone=PolycomSoundPointIP-SPIP_550-UA/3.3.2.0413”) in new stack
– Executing [[email protected]:5] ExecIf(“Local/[email protected];2”, “0?Set(CALLINFO=Call-Info: sip:broadworks.net;answer-after=0)”) in new stack
– Executing [[email protected]:6] ExecIf(“Local/[email protected];2”, “0?Set(ALERTINFO=Alert-Info: Intercom)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:7] ExecIf(“Local/[email protected];2”, “1?SipAddHeader(Alert-Info: Ring Answer)”) in new stack
– Executing [[email protected]:8] ExecIf(“Local/[email protected];2”, “1?SipAddHeader(Call-Info: ;answer-after=0)”) in new stack
– Called 104
– Executing [[email protected]:9] ExecIf(“Local/[email protected];2”, “1?Set(__SIP_URI_OPTIONS=intercom=true)”) in new stack
– Executing [[email protected]:2] Dial(“Local/[email protected];2”, “SIP/303,5,A(beep)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called 303
– <SIP/105-00000127>AGI Script page.agi completed, returning 0
– Executing [[email protected]:11] ConfBridge(“SIP/105-00000127”, “1327607975480,qwAG,”) in new stack
– SIP/301-00000128 is ringing
– SIP/102-00000129 is ringing
– SIP/104-0000012b is ringing
– SIP/103-0000012a is ringing
– SIP/303-0000012c is ringing

-- Executing [[email protected]:2] Answer("Local/[email protected];2", "") in new stack
-- Executing [[email protected]:3] ConfBridge("Local/[email protected];2", "1327607975480,,,") in new stack
-- <Local/[email protected];1> Playing 'beep.gsm' (language 'en')








-- SIP/301-00000128 answered Local/[email protected];2
-- <SIP/301-00000128> Playing 'beep.gsm' (language 'en')
-- SIP/102-00000129 answered Local/[email protected];2
-- <SIP/102-00000129> Playing 'beep.gsm' (language 'en')
-- SIP/104-0000012b answered Local/[email protected];2
-- <SIP/104-0000012b> Playing 'beep.gsm' (language 'en')
-- SIP/103-0000012a answered Local/[email protected];2
-- <SIP/103-0000012a> Playing 'beep.gsm' (language 'en')
-- SIP/303-0000012c answered Local/[email protected];2
-- <SIP/303-0000012c> Playing 'beep.gsm' (language 'en')

== Spawn extension (app-paging, PAGE301, 2) exited non-zero on ‘Local/[email protected];2’
== Spawn extension (app-paging, PAGE102, 2) exited non-zero on ‘Local/[email protected];2’
== Spawn extension (app-paging, PAGE104, 2) exited non-zero on ‘Local/[email protected];2’
== Spawn extension (app-paging, PAGE303, 2) exited non-zero on ‘Local/[email protected];2’
== Spawn extension (app-paging, PAGE103, 2) exited non-zero on ‘Local/[email protected];2’
– Executing [[email protected]:1] ExecIf(“SIP/105-00000127”, “1?Set(DEVICE_STATE(Custom:PAGE601)=NOT_INUSE)”) in new stack

I also got the same under Asterisk 1.8.7.1 and FreePBX 2.10.0 rc1.0 w/latest updates.

It appears that Paging and Intercom v2.10.0.2 is bugged.

Before the updates a day or so ago, paging and intercom worked just fine.

The bug also entails having to reset the PBX server since the audio channels are hung and the phones wont hang up or wont reister if hard reset.

Kinda embarassing/funny scenario to be honest, especially when you have 20+ phones connected. This sort of thing happens I suppose when you use a beta, so no love lost here.

Thanks Devs!

Are you using meet-me or app-conference for pages. I know it seems app-conference does not allow hanging up the channels with newer Asterisk 1.8 systems

Just the Paging and Intercom v2.10.0.2 nothing else.

No paging requires meet-me or app_conference. You need to tell us which one you are using.

Sorry I looked but nothing tells me the information you are requesting.
Where do I look for this or can you just assume I am using whatever the default is?
I did not at any time select or choose an app either way that I am aware of so… I dunno.

Conference Room App = app_meetme <<<---- ???

I hunted everywhere for the info and this is the only thing I could find that referenced app_meetme or app_conference. Not sure what that has to do with paging though.

app_meetme and app_conference are important to paging because paging is accomplished by putting all the phones you are calling in a conference call.

So does this setting define and gives the answer Tony was looking for?
Conference Room App = app_meetme

I have disabled intercom & paging for now so not a big deal, just wanted to report the issue and get a solution. It was working before the recent updates, and nothing else has changed in my configuration.

Does anyone else share this problem?

@elihunter - as per your call trace you are using app_confbridge. In * < 10, app_confbridge does not offer the option to hang up the page when the originator of the page hangs up. This is an Asterisk limitation.

I have the same issue at one installation, but it’s intermittent. i THINK it may happen when someone hit the spped dial for a page group and then hangs up too fast… haven’t confirmed this yet.

Same issue here. After paging a group consisting of 40 phones, on average 8 of the phones will not hang up after the page. This results in the phone being off-hook until the user presses “end” or “cancel” to cut that channel.

Channel Location State Application(Data)
SIP/250-00000886 [email protected]:1 Up MeetMe(1334329551168,1doqsxm,
SIP/241-000009b9 [email protected] Ring Dial(SIP/VMS/4169799996,300,tT
SIP/237-00000880 [email protected]:1 Up MeetMe(1334329551168,1doqsxm,
SIP/VMS-000009ba [email protected] Down AppDial((Outgoing Line))
DAHDI/pseudo-5180299 [email protected]:1 Rsrvd (None)
SIP/353-00000894 [email protected]:1 Up MeetMe(1334329551168,1doqsxm,
SIP/232-0000087a [email protected]:1 Up MeetMe(1334329551168,1doqsxm,
SIP/252-0000088a [email protected]:1 Up MeetMe(1334329551168,1doqsxm,
SIP/204-00000878 [email protected]:1 Up MeetMe(1334329551168,1doqsxm,
SIP/236-00000883 [email protected]:1 Up MeetMe(1334329551168,1doqsxm,
10 active channels
1 active call

Is there some setting that i can make to allow paging to work as expected, that is, have all phones answer when paged (working) and when the pager hangs up, the pagees’ phones will also hang up.

Frankly, 40 phones is a bit much. If the phones support multicast paging I would use that.

Has anyone resolved this? We’re seeing the same issue with a 20-phone system, all Aastra 6757i units, on Asterisk Ver. 1.8.14.0 and FreePBX 2.10.1.1. We didn’t see this behavior before upgrading from Asterisk 1.6 (which had more severe issues of its own).

After a group page is performed at one extension, all phones appear to be left “off-hook”, and after pressing “Good Bye”, they cannot page or perform transfers.

No responses here, so I’m going to try updating the firmware on the 57i units and changing the paging soft key as follows:

paging group listening: 224.0.0.2:10000,239.0.1.20:15000
softkey1 type: paging
softkey1 label: ”Group 1”
softkey1 value: 224.0.0.2:10000

This is what the Aastra Admin Guide uses as an example. Any advice when doing this?

Sorry to revive a dead thread, but has this been resolved at all? i am using the FreePBX Distro with FreePBX 2.210.62-5 and whichever Asterisk it comes with. I am using a polycom 335, 550, and 650.

My settings are set to use app-meetme as the conference app, but i noticed the page.agi uses the app-conference to make the pages. Single phone paging works fine, but when dialing a page group, the paged phones will stay connected until hanging up manually. they do not tie up a channel and can be used normally right away after hanging up, but having users manually hang up after a page is performed is annoying.

With FreePBX 2.11.0.0rc1.2 we are seeing intermittent “no hang up” issues with paging as well.

I’m at a loss as to how to resolve this problem.

We never resolved it. Little quirks like these that never get fixed are annoying. Worse is that our company is now looking an microsoft lync to replace freepbx.