Paging Group disconnecting immediately in FreePBX 2.9.0.9

Hi all!

I have a problem with my paging group. The paging group is 112, and the only extension in the group (currently) is Extension 209. In reality, I have about a dozen extensions in my paging group, but for this experiment, I narrowed it down to one (209) to try to simplify things.

Here’s the problem. I dial 112 from my phone (Grandstream GXP-2000, if it matters). It makes the paging tone, and all of the phones in the paging group ring and make the tone, as if they are answering a page.

For some reason, though, it disconnects right after this. Basically, I dial 112, it makes the tone, and then it’s disconnected.

I have also tried this from a different phone, just in case it was something to do with the Grandstreams.

I CAN directly page an extension. For instance, if I dial 209, it will automatically pick up and I can talk. So that works just fine.

I ran a “tail -f /var/log/asterisk/full” in hopes that it would provide the answer. Here’s what I have:

[2012-02-21 09:30:25] VERBOSE[9518] netsock2.c: == Using SIP RTP TOS bits 184
[2012-02-21 09:30:25] VERBOSE[9518] netsock2.c: == Using SIP RTP CoS mark 5
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:1] Answer(“SIP/211-000001f6”, “”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:2] Macro(“SIP/211-000001f6”, “user-callerid,”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:1] Set(“SIP/211-000001f6”, “AMPUSER=211”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:2] GotoIf(“SIP/211-000001f6”, “0?report”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/211-000001f6”, “1?Set(REALCALLERIDNUM=211)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:4] Set(“SIP/211-000001f6”, “AMPUSER=211”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:5] Set(“SIP/211-000001f6”, “AMPUSERCIDNAME=Luke”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:6] GotoIf(“SIP/211-000001f6”, “0?report”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:7] Set(“SIP/211-000001f6”, “AMPUSERCID=211”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:8] Set(“SIP/211-000001f6”, “CALLERID(all)=“Luke” <211>”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:9] GotoIf(“SIP/211-000001f6”, “0?limit”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:10] ExecIf(“SIP/211-000001f6”, “0?Set(GROUP(concurrency_limit)=211)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:11] ExecIf(“SIP/211-000001f6”, “0?Set(CHANNEL(language)=)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:12] GotoIf(“SIP/211-000001f6”, “0?continue”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:13] Set(“SIP/211-000001f6”, “__TTL=64”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:14] GotoIf(“SIP/211-000001f6”, “1?continue”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Goto (macro-user-callerid,s,25)
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:25] Set(“SIP/211-000001f6”, “CALLERID(number)=211”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:26] Set(“SIP/211-000001f6”, “CALLERID(name)=Luke”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:27] Set(“SIP/211-000001f6”, “CHANNEL(language)=en”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:3] Set(“SIP/211-000001f6”, “_AMPUSER=211”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:4] Set(“SIP/211-000001f6”, “_SIPURI=”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:5] Set(“SIP/211-000001f6”, “_ALERTINFO=Alert-Info: Ring Answer”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:6] Set(“SIP/211-000001f6”, “_CALLINFO=Call-Info: ;answer-after=0”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:7] Set(“SIP/211-000001f6”, “_SIPURI=intercom=true”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:8] Set(“SIP/211-000001f6”, “_DOPTIONS=A(beep)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:9] Set(“SIP/211-000001f6”, “_DTIME=5”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:10] Set(“SIP/211-000001f6”, “_ANSWERMACRO=”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:11] Set(“SIP/211-000001f6”, “__FORWARD_CONTEXT=block-cf”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:12] Page(“SIP/211-000001f6”, “LOCAL/[email protected]”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] dial.c: – Called [email protected]
[2012-02-21 09:30:25] VERBOSE[21987] file.c: – <SIP/211-000001f6> Playing ‘beep.gsm’ (language ‘en’)
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:1] GotoIf(“Local/[email protected];2”, “0?skipself”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:2] ChanIsAvail(“Local/[email protected];2”, “SIP/209,s”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] netsock2.c: == Using SIP RTP TOS bits 184
[2012-02-21 09:30:25] VERBOSE[21988] netsock2.c: == Using SIP RTP CoS mark 5
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:3] GotoIf(“Local/[email protected];2”, “0?skipself”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:4] GotoIf(“Local/[email protected];2”, “0?skipself”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:5] Macro(“Local/[email protected];2”, “autoanswer,209”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:1] Set(“Local/[email protected];2”, “DIAL=SIP/209”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:2] ExecIf(“Local/[email protected];2”, “0?Set(DIAL=DAHDI/209)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:3] GotoIf(“Local/[email protected];2”, “0?macro”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:4] Set(“Local/[email protected];2”, “phone=Grandstream GXP2000 1.2.5.3”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:5] ExecIf(“Local/[email protected];2”, “0?Set(CALLINFO=Call-Info: sip:broadworks.net;answer-after=0)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:6] ExecIf(“Local/[email protected];2”, “1?SipAddHeader(Alert-Info: Ring Answer)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:7] ExecIf(“Local/[email protected];2”, “1?SipAddHeader(Call-Info: ;answer-after=0)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:8] ExecIf(“Local/[email protected];2”, “1?Set(__SIP_URI_OPTIONS=intercom=true)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: – Executing [[email protected]:6] Dial(“Local/[email protected]xt-paging-97e9;2”, “SIP/209,5,A(beep)”) in new stack
[2012-02-21 09:30:25] VERBOSE[21988] netsock2.c: == Using SIP RTP TOS bits 184
[2012-02-21 09:30:25] VERBOSE[21988] netsock2.c: == Using SIP RTP CoS mark 5
[2012-02-21 09:30:25] VERBOSE[21988] app_dial.c: – Called SIP/209
[2012-02-21 09:30:25] VERBOSE[21988] app_dial.c: – SIP/209-000001f8 is ringing
[2012-02-21 09:30:25] VERBOSE[21989] dial.c: – Local/[email protected];1 is ringing
[2012-02-21 09:30:25] WARNING[21987] app_meetme.c: Unable to open DAHDI pseudo device
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: == Spawn extension (from-internal, 112, 12) exited non-zero on ‘SIP/211-000001f6’
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/211-000001f6”, “”) in new stack
[2012-02-21 09:30:25] VERBOSE[21987] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/211-000001f6’
[2012-02-21 09:30:25] VERBOSE[21988] pbx.c: == Spawn extension (ext-paging, PAGE209, 6) exited non-zero on ‘Local/[email protected];2’
[2012-02-21 09:30:25] WARNING[9518] chan_sip.c: Unsupported SDP media type in offer: video 0 RTP/AVP 31 34 98 99


I see those two warnings in the bottom several lines, but I’m not entirely sure what they mean. I am not using video, and I don’t use any DAHDI hardware.

Thank you!

Luke

“I don’t use any DAHDI hardware” <---- that is your issue! app_meetme requires dahdi as timing source, no dahdi means no app_meetme.

Thanks for the reply!

I should clarify that I don’t THINK I use DAHDI – that is, I don’t think I ever configured it. I just have a voip.ms trunk that I use, and have no special hardware cards or anything in my FreePBX system.

So, with hopefully not sounding too dumb, I guess I wonder what I need to do to get it working again? Like I said, it did work previously, but that was before I moved the FreePBX installation to a different server – with different hardware.

Thanks again!

After doing a “service dahdi restart,” things worked again!

Not sure why it went out in the first place, but I guess I’ll keep an eye on it.

Any thoughts as to why it would need restarting? Maybe just a fluke?

Still having this intermittent problem! When the problem occurs, it happens to ALL paging groups. They all disconnect just a second after the “beep.”

Just wondering if anybody has ever encountered this/solved it!

This isn’t a FreePBX issue. It’s Asterisk and DAHDI drivers.

If you don’t have any hardware you need to configure DAHDI dummy timing source.

http://docs.tzafrir.org.il/dahdi-tools/README.html

More than likely dahdi is already install on your system and you just need to configure the timing to recognize your new hardware.

http://docs.tzafrir.org.il/dahdi-tools/man/dahdi_genconf.8.html

Thanks for the info.

I don’t have any DAHDI hardware – this is straight VoIP.

From those links you shared, I can’t really tell how to configure the dummy timing source. This thread suggested I comment out all the drivers from /etc/dahdi/modules: http://www.selbytech.com/2010/01/how-to-setup-asterisk-1-6-2-on-centos-5-4/

I did that, saved the modules file, and restarted the DAHDI service. Things seem to work okay, but I don’t know if the problem is fixed. I would guess it’s not, because I haven’t done anything with the dummy timing source. I’m trying to figure out what that means or how it’s done.

Use the ‘dahdi show channels’ in Asterisk.

It will look like this:

[[email protected] ~]# asterisk -r
maieast*CLI> dahdi show channels
   Chan Extension  Context         Language   MOH Interpret        Blocked    State
 pseudo            default                    default                         In Service
maieast*CLI>

Do a ‘service dahdi status’ from CentOS and see if the dummy driver is loaded.

Then do an amportal stop, dahdi_gencof -vvvv -F, then amportal start.

Check your dahdi channels in Asterisk and you should be all set.

Here’s what I get:

pbx*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service

[email protected]:/etc/asterisk $ service dahdi status
[email protected]:/etc/asterisk $

[email protected]:/etc/asterisk $ dahdi_genconf -vvvv -F
Default parameters from /etc/dahdi/genconf_parameters
Empty configuration – no spans
Generating /etc/dahdi/system.conf
Empty configuration – no spans
Generating /etc/asterisk/dahdi-channels.conf

I ran dahdi_genconf -vvvv -F after stopping Asterisk, and restarted it afterwards. Does the blank output of “service dahdi status” tell us anything?

Thanks for your help!

Luke