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Paging Pro: No sound on Multicast Page


(Steve) #1

I’m trying to get multicast paging working with the Paging Pro commercial module and Grandstream GXP2130 phones. The 2130s support multicast paging and it works if I set one of the 2130’s programmable buttons to “Multicast Paging” mode and set a multicast address and port.

When I press the programmed button, other phones set to listen this multicast address and port automatically activate a line button and play the audio over the speaker. So far so good.

I can’t for the life of me get this working using the Paging Pro module. In the Paging and Intercom module I have created a new Page Group named “Page All”. I have the following settings:

Paging Extension: 100
Group Description: Page All
Device List: (empty)

Busy Extensions: Skip
Duplex: (unchecked)
Default Page Group: (unchecked)

Busy Page Group: Do Nothing
Page Announce: Default
CID Prepend: (blank)
RTP Multicast: 239.1.1.1:6767 (this is the same IP address and port used on the 2130’s “Multicast Listening” config section)

When I dial extension 100 I see all of the phones react; their line 1 LEDs light and the LCDs show “Page All” but I get no audio from the phones. Any idea what I might be doing wrong here?


(Andrew Nagy) #2

You are using FreePBX 13 and Asterisk 13 right?


(Steve) #3

I’m using Asterisk 13.2, and FreePBX 12.0.50.1. I knew Asterisk 13 was a requirement but didn’t realize FreePBX 13 was required. Does it not work at all with 12.0.50.1? If not, how stable is the current FreePBX Alpha? I’m setting this up for our special needs school and we need Page All functionality for 90+ extensions. So regular page groups isn’t an option.

Thanks – Steve


(Rob Thomas) #4

This is a Codec issue. Asterisk is sending audio to the phone in a Codec that the phone doesn’t understand. I’ll chase up the bug report for that and see if I can figure out what happened.

Edit, 1 hour later: Apparently they changed from ‘Codec’ to ‘Codecs’. I’ve just released Paging 12.0.14 that fixes this. Upgrade and everything should start working happily!

–Rob


(Steve) #5

Wow that was fast. I’ll give it a try in the morning. Thanks!


(Steve) #6

Rob,

Paging v12.0.14 fixes the no audio issue but the audio quality is poor and almost unintelligible with what sounds like very rapid interval drop outs for lack of a better description. By comparison, the audio quality is fine if I program a button for “Multicast Paging” mode using the phone’s web GUI and I initiate the multicast page using the programmed button.

Looking at the available codecs in the multicast paging section of the Grandstream phone web GUI, I have the following “Multicast Paging Codec” options:

G.729A/B
PCMU (this is the default)
PCMA
G.726-32
G.722 (wide band)

These settings only affect the codec used when a page is initiated using the programmed button set to “Multicast Paging” mode referenced above. Is it possible the phone doesn’t like the codec that FreePBX/Asterisk is using?


(Rob Thomas) #7

The codec it’s using for multicast paging is G711u (ulaw) - or it should be.

You can verify this with ‘core show channel Multi[PUSH TAB]’ when you’re paging, and you should see ‘Codec: ulaw’ about half way down the list.

This is what I see:
State: Up (6)
NativeFormats: (ulaw)
WriteFormat: ulaw
ReadFormat: ulaw
WriteTranscode: No
ReadTranscode: No

Edit: Just re-tested it on a couple of SPA504s and it worked fine, with perfect audio.


(Steve) #8

Sorry, not sure I follow. I’m attempting to do this remotely using an SSH session in Putty.

I’m entering the Asterisk CLI using: ‘asterisk -vvvvvvvvvvr’

At the *CLI prompt, should I enter '‘core show channel Multi[PUSH TAB]’ or ‘core show channel Multi’ and press the tab key.

When I do the former I get this response:

Usage: core show channel
Shows lots of information about the specified channel.

If I do the latter, pressing the tab key does nothing.

Anyway, assuming that the command is ‘core show channel Multi[PUSH TAB]’, I schedule a page to occur a couple of minutes into the future and wait. When the time elapses the console displays the follow:

BedfordPBX*CLI>
[2015-04-01 17:21:01] ERROR[9255]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2015-04-01 17:21:01] ERROR[9255]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
– Attempting call on Local/101@from-internal for application Playback(custom/AllStaffToBusses) (Retry 1)
– Executing [101@from-internal:1] Goto(“Local/101@from-internal-0000000c;2”, “app-pagegroups,101,1”) in new stack
– Called 101@from-internal
– Goto (app-pagegroups,101,1)
– Executing [101@app-pagegroups:1] Set(“Local/101@from-internal-0000000c;2”, “MCAST=239.1.1.1:6767”) in new stack
– Executing [101@app-pagegroups:2] Macro(“Local/101@from-internal-0000000c;2”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“Local/101@from-internal-0000000c;2”, “TOUCH_MONITOR=1427923261.232”) in new stack
– Executing [s@macro-user-callerid:2] Set(“Local/101@from-internal-0000000c;2”, “AMPUSER=101”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“Local/101@from-internal-0000000c;2”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“Local/101@from-internal-0000000c;2”, “1?Set(REALCALLERIDNUM=101)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“Local/101@from-internal-0000000c;2”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“Local/101@from-internal-0000000c;2”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“Local/101@from-internal-0000000c;2”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“Local/101@from-internal-0000000c;2”, “1?report”) in new stack
– Goto (macro-user-callerid,s,16)
– Executing [s@macro-user-callerid:16] GotoIf(“Local/101@from-internal-0000000c;2”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:17] ExecIf(“Local/101@from-internal-0000000c;2”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
– Executing [s@macro-user-callerid:18] Set(“Local/101@from-internal-0000000c;2”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:19] GotoIf(“Local/101@from-internal-0000000c;2”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,30)
– Executing [s@macro-user-callerid:30] Set(“Local/101@from-internal-0000000c;2”, “CALLERID(number)=101”) in new stack
– Executing [s@macro-user-callerid:31] Set(“Local/101@from-internal-0000000c;2”, “CALLERID(name)=All Staff To Busses”) in new stack
– Executing [s@macro-user-callerid:32] Set(“Local/101@from-internal-0000000c;2”, “CDR(cnum)=101”) in new stack
– Executing [s@macro-user-callerid:33] Set(“Local/101@from-internal-0000000c;2”, “CDR(cnam)=All Staff To Busses”) in new stack
– Executing [s@macro-user-callerid:34] Set(“Local/101@from-internal-0000000c;2”, “CHANNEL(language)=en”) in new stack
– Executing [101@app-pagegroups:3] Set(“Local/101@from-internal-0000000c;2”, “_PAGEGROUP=101”) in new stack
– Executing [101@app-pagegroups:4] GotoIf(“Local/101@from-internal-0000000c;2”, “1?:busy”) in new stack
– Executing [101@app-pagegroups:5] Set(“Local/101@from-internal-0000000c;2”, “DEVICE_STATE(Custom:PAGE101)=INUSE”) in new stack
– Executing [101@app-pagegroups:6] Gosub(“Local/101@from-internal-0000000c;2”, “app-paging,ssetup,1()”) in new stack
– Executing [ssetup@app-paging:1] Set(“Local/101@from-internal-0000000c;2”, “_SIPURI=”) in new stack
– Executing [ssetup@app-paging:2] Set(“Local/101@from-internal-0000000c;2”, “_ALERTINFO=Ring Answer”) in new stack
– Executing [ssetup@app-paging:3] Set(“Local/101@from-internal-0000000c;2”, “_CALLINFO=;answer-after=0”) in new stack
– Executing [ssetup@app-paging:4] Set(“Local/101@from-internal-0000000c;2”, “_SIPURI=intercom=true”) in new stack
– Executing [ssetup@app-paging:5] Set(“Local/101@from-internal-0000000c;2”, “_DTIME=5”) in new stack
– Executing [ssetup@app-paging:6] Set(“Local/101@from-internal-0000000c;2”, “_ANSWERMACRO=”) in new stack
– Executing [ssetup@app-paging:7] Set(“Local/101@from-internal-0000000c;2”, “PAGE_CONF=1427923261496”) in new stack
– Executing [ssetup@app-paging:8] Return(“Local/101@from-internal-0000000c;2”, “”) in new stack
– Executing [101@app-pagegroups:7] Set(“Local/101@from-internal-0000000c;2”, “PAGEMODE=PAGE”) in new stack
– Executing [101@app-pagegroups:8] Set(“Local/101@from-internal-0000000c;2”, “PAGE_MEMBERS=”) in new stack
– Executing [101@app-pagegroups:9] Set(“Local/101@from-internal-0000000c;2”, “PAGE_CONF_OPTS=”) in new stack
– Executing [101@app-pagegroups:10] ExecIf(“Local/101@from-internal-0000000c;2”, “0?Set(STREAM=custom/AllStaffToBusses):Set(STREAM=NONE)”) in new stack
– Executing [101@app-pagegroups:11] AGI(“Local/101@from-internal-0000000c;2”, “page.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/page.agi
– Called s@app-page-stream
– Executing [s@app-page-stream:1] Wait(“Local/s@app-page-stream-0000000d;2”, “1”) in new stack
– Called PAGErtp@app-paging/n
– Executing [PAGErtp@app-paging:1] Macro(“Local/PAGErtp@app-paging-0000000e;2”, “autoanswer,rtp”) in new stack
– Executing [s@macro-autoanswer:1] GotoIf(“Local/PAGErtp@app-paging-0000000e;2”, “1?knowndial”) in new stack
– Goto (macro-autoanswer,s,19)
– Executing [s@macro-autoanswer:19] Set(“Local/PAGErtp@app-paging-0000000e;2”, “DIAL=MulticastRTP/basic/239.1.1.1:6767”) in new stack
– Executing [s@macro-autoanswer:20] ExecIf(“Local/PAGErtp@app-paging-0000000e;2”, “0?Set(DIAL=DAHDIticastRTP/basic/239.1.1.1:6767)”) in new stack
– Executing [s@macro-autoanswer:21] GotoIf(“Local/PAGErtp@app-paging-0000000e;2”, “0?macro”) in new stack
– Executing [s@macro-autoanswer:22] Set(“Local/PAGErtp@app-paging-0000000e;2”, “USERAGENT=”) in new stack
– Executing [s@macro-autoanswer:23] ExecIf(“Local/PAGErtp@app-paging-0000000e;2”, “1?Set(USERAGENT=rtp)”) in new stack
– Executing [s@macro-autoanswer:24] ExecIf(“Local/PAGErtp@app-paging-0000000e;2”, “0?Set(CALLINFO=sip:broadworks.net;answer-after=0)”) in new stack
– Executing [s@macro-autoanswer:25] ExecIf(“Local/PAGErtp@app-paging-0000000e;2”, “0?Set(ALERTINFO=Intercom)”) in new stack
– Executing [s@macro-autoanswer:26] ExecIf(“Local/PAGErtp@app-paging-0000000e;2”, “0?Set(ALERTINFO=info=Auto Answer)”) in new stack
– Executing [s@macro-autoanswer:27] ExecIf(“Local/PAGErtp@app-paging-0000000e;2”, “0?Set(ALERTINFO=ring-answer)”) in new stack
– Executing [s@macro-autoanswer:28] ExecIf(“Local/PAGErtp@app-paging-0000000e;2”, “1?Set(__SIP_URI_OPTIONS=intercom=true)”) in new stack
– Executing [PAGErtp@app-paging:2] Set(“Local/PAGErtp@app-paging-0000000e;2”, “_DOPTIONS=A(beep)b(autoanswer^s^1(Ring Answer,;answer-after=0))”) in new stack
– Executing [PAGErtp@app-paging:3] Dial(“Local/PAGErtp@app-paging-0000000e;2”, “MulticastRTP/basic/239.1.1.1:6767,5,A(beep)b(autoanswer^s^1(Ring Answer,;answer-after=0))”) in new stack
– MulticastRTP/0x7f542c004188 Internal Gosub(autoanswer,s,1(Ring Answer,;answer-after=0)) start
– Executing [s@autoanswer:1] GosubIf(“MulticastRTP/0x7f542c004188”, “1?addheader,1(Alert-Info,Ring Answer)”) in new stack
– Executing [addheader@autoanswer:1] SIPAddHeader(“MulticastRTP/0x7f542c004188”, “Alert-Info: Ring Answer”) in new stack
– Executing [addheader@autoanswer:2] Set(“MulticastRTP/0x7f542c004188”, “PJSIP_HEADER(add,Alert-Info)=Ring Answer”) in new stack
[2015-04-01 17:21:02] ERROR[9300][C-0000000f]: pbx.c:4427 ast_func_write: Function PJSIP_HEADER not registered
– Executing [addheader@autoanswer:3] Return(“MulticastRTP/0x7f542c004188”, “”) in new stack
– Executing [s@autoanswer:2] GosubIf(“MulticastRTP/0x7f542c004188”, “1?addheader,1(Call-Info,;answer-after=0)”) in new stack
– Executing [addheader@autoanswer:1] SIPAddHeader(“MulticastRTP/0x7f542c004188”, “Call-Info: ;answer-after=0”) in new stack
– Executing [addheader@autoanswer:2] Set(“MulticastRTP/0x7f542c004188”, “PJSIP_HEADER(add,Call-Info)=;answer-after=0”) in new stack
[2015-04-01 17:21:02] ERROR[9300][C-0000000f]: pbx.c:4427 ast_func_write: Function PJSIP_HEADER not registered
– Executing [addheader@autoanswer:3] Return(“MulticastRTP/0x7f542c004188”, “”) in new stack
– Executing [s@autoanswer:3] Return(“MulticastRTP/0x7f542c004188”, “”) in new stack
== Spawn extension (default, PAGErtp, 1) exited non-zero on ‘MulticastRTP/0x7f542c004188’
– MulticastRTP/0x7f542c004188 Internal Gosub(autoanswer,s,1(Ring Answer,;answer-after=0)) complete GOSUB_RETVAL=
– <Local/101@from-internal-0000000c;2>AGI Script page.agi completed, returning 0
– Called MulticastRTP/basic/239.1.1.1:6767
– MulticastRTP/0x7f542c004188 answered Local/PAGErtp@app-paging-0000000e;2
– <MulticastRTP/0x7f542c004188> Playing ‘beep.ulaw’ (language ‘en’)
– Executing [101@app-pagegroups:12] Set(“Local/101@from-internal-0000000c;2”, “CONFBRIDGE(user,template)=page_user_duplex”) in new stack
– Executing [101@app-pagegroups:13] Set(“Local/101@from-internal-0000000c;2”, “CONFBRIDGE(user,admin)=yes”) in new stack
– Executing [101@app-pagegroups:14] Set(“Local/101@from-internal-0000000c;2”, “CONFBRIDGE(user,marked)=yes”) in new stack
– Executing [101@app-pagegroups:15] Answer(“Local/101@from-internal-0000000c;2”, “”) in new stack
– Local/101@from-internal-0000000c;1 answered
> Launching Playback(custom/AllStaffToBusses) on Local/101@from-internal-0000000c;1
– <Local/101@from-internal-0000000c;1> Playing ‘custom/AllStaffToBusses.slin’ (language ‘en’)
– Executing [101@app-pagegroups:16] ConfBridge(“Local/101@from-internal-0000000c;2”, “1427923261496,admin_menu”) in new stack
– Channel Local/101@from-internal-0000000c;2 joined ‘softmix’ base-bridge <28a04a2f-12c1-4ff2-a081-b7570651a514>
– Local/PAGErtp@app-paging-0000000e;1 answered
> Launching ConfBridge(1427923261496,user_menu) on Local/PAGErtp@app-paging-0000000e;1
– Channel Local/PAGErtp@app-paging-0000000e;2 joined ‘simple_bridge’ basic-bridge <7c3c1845-5d8d-4c40-86b3-2a396a65efc8>
– Channel Local/PAGErtp@app-paging-0000000e;1 joined ‘softmix’ base-bridge <28a04a2f-12c1-4ff2-a081-b7570651a514>
– Channel MulticastRTP/0x7f542c004188 joined ‘simple_bridge’ basic-bridge <7c3c1845-5d8d-4c40-86b3-2a396a65efc8>
– Executing [s@app-page-stream:2] Answer(“Local/s@app-page-stream-0000000d;2”, “”) in new stack
– Local/s@app-page-stream-0000000d;1 answered
> Launching Wait(5) on Local/s@app-page-stream-0000000d;1
– Executing [s@app-page-stream:3] Set(“Local/s@app-page-stream-0000000d;2”, “CONFBRIDGE(user,template)=page_user_duplex”) in new stack
– Executing [s@app-page-stream:4] Set(“Local/s@app-page-stream-0000000d;2”, “CONFBRIDGE(user,marked)=yes”) in new stack
– Executing [s@app-page-stream:5] ConfBridge(“Local/s@app-page-stream-0000000d;2”, “1427923261496,”) in new stack
– Channel Local/s@app-page-stream-0000000d;2 joined ‘softmix’ base-bridge <28a04a2f-12c1-4ff2-a081-b7570651a514>
– Channel Local/101@from-internal-0000000c;2 left ‘softmix’ base-bridge <28a04a2f-12c1-4ff2-a081-b7570651a514>
[2015-04-01 17:21:06] NOTICE[9281]: pbx_spool.c:401 attempt_thread: Call completed to Local/101@from-internal
– Executing [h@app-pagegroups:1] ExecIf(“Local/101@from-internal-0000000c;2”, “1?Set(DEVICE_STATE(Custom:PAGE101)=NOT_INUSE)”) in new stack
– Executing [h@app-pagegroups:2] GosubIf(“Local/101@from-internal-0000000c;2”, “0?record-page,1()”) in new stack
– Executing [h@app-pagegroups:3] ExecIf(“Local/101@from-internal-0000000c;2”, “0?System(rm .sln)”) in new stack
– Executing [h@app-pagegroups:4] ExecIf(“Local/101@from-internal-0000000c;2”, “0?System(rm -f /var/spool/asterisk/outgoing/)”) in new stack
– Channel Local/s@app-page-stream-0000000d;2 left ‘softmix’ base-bridge <28a04a2f-12c1-4ff2-a081-b7570651a514>
– Channel Local/PAGErtp@app-paging-0000000e;1 left ‘softmix’ base-bridge <28a04a2f-12c1-4ff2-a081-b7570651a514>
– Channel Local/PAGErtp@app-paging-0000000e;2 left ‘simple_bridge’ basic-bridge <7c3c1845-5d8d-4c40-86b3-2a396a65efc8>
– Channel MulticastRTP/0x7f542c004188 left ‘simple_bridge’ basic-bridge <7c3c1845-5d8d-4c40-86b3-2a396a65efc8>
== Spawn extension (app-paging, PAGErtp, 3) exited non-zero on ‘Local/PAGErtp@app-paging-0000000e;2’

Sorry for the newbieness and thanks for the help – Steve


(Rob Thomas) #9

You should type ‘core show channel Multi’ and then push the tab key.

Then there is no multicast stream happening. Try ‘core show channels’ to see which channels are active. You should see something like this:

ssl*CLI> core show channels Channel Location State Application(Data) Local/PAGErtp@app-pa (None) Up ConfBridge(1427927059629,,,use Local/PAGErtp@app-pa (None) Up Dial(MulticastRTP/basic/224.16 MulticastRTP/0x7fa03 (None) Up AppDial((Outgoing Line)) SIP/304-00000002 (None) Up ConfBridge(1427927059629,,,adm 4 active channels 2 active calls 7 calls processed ssl*CLI>

I have a sneaking suspicion that you’re NOT using multicast, which would explain the terrible audio?

Edit: I notice in the log above you ARE using Multicast, so it’s got me stumped.


(Steve) #10

I should note that the console beeps when I press the tab key indicating that the key press isn’t valid(??). I’m not on-site ATM I’ll check again in the morning.

Thanks – Steve


(Andrew Nagy) #12

You need to do this while you are paging.


(Steve) #13

Operator error; I was hitting tab before the page started.

Here’s the console output after running ‘core show channel Multi’ and pressing the tab key. Looks like it is using ulaw:

BedfordPBX*CLI> core show channel MulticastRTP/0x7f542c003308
– General –
Name: MulticastRTP/0x7f542c003308
Type: MulticastRTP
UniqueID: 1427974774.366
LinkedID: 1427974774.358
Caller ID: PAGErtp
Caller ID Name: Ed Rego
Connected Line ID: 3230
Connected Line ID Name: Ed Rego
Eff. Connected Line ID: 3230
Eff. Connected Line ID Name: Ed Rego
DNID Digits: (N/A)
Language: en
State: Up (6)
NativeFormats: (ulaw)
WriteFormat: ulaw
ReadFormat: ulaw
WriteTranscode: No
ReadTranscode: No
Time to Hangup: 0
Elapsed Time: 0h0m17s
Bridge ID: dd4b4c08-3850-4dec-a53b-8252d12581f3
– PBX –
Context: default
Extension:
Priority: 1
Call Group: 0
Pickup Group: 0
Application: AppDial
Data: (Outgoing Line)
Call Identifer: [C-00000017]
Variables:
BRIDGEPEER=Local/PAGErtp@app-paging-00000013;2
GOSUB_RETVAL=
SIPADDHEADER02=Call-Info: ;answer-after=0
SIPADDHEADER01=Alert-Info: Ring Answer
DIALEDPEERNUMBER=basic/239.1.1.1:6767
DOPTIONS=A(beep)b(autoanswer^s^1(Ring Answer,;answer-after=0))
SIP_URI_OPTIONS=intercom=true
FORWARD_CONTEXT=block-cf
CDR Variables:
level 1: calledsubaddr=
level 1: callingsubaddr=
level 1: dnid=
level 1: clid="Ed Reg
level 1: src=PAGErtp
level 1: dcontext=default
level 1: channel=Multica
level 1: lastapp=AppDial
level 1: lastdata=(Outgoi
level 1: start=1427974
level 1: answer=1427974
level 1: end=1427974
level 1: duration=0
level 1: billsec=0
level 1: disposition=8
level 1: amaflags=3
level 1: uniqueid=1427974
level 1: linkedid=1427974
level 1: sequence=61


(Steve) #14

Any other thoughts? I bought 100 Grandstream GXP2130s and we’re scheduled to replace our existing Nortel system during a school vacation break during the week of 4/19. Unfortunately, that’s not gonna happen if I can’t get page all working.

Thanks – Steve


(Rob Thomas) #15

Sorry, no. You can mess around with the codec yourself, if you think it’s that - you can edit the file /var/lib/asterisk/agi-bin/page.agi and the codec that’s used for multicast paging is on line 247.

I’d ask the people you bought them from for assistance., if you can’t figure it out. (If you bought them from Sangoma/Schmooze, then we can probably make this pretty easy)


(Steve) #16

All 100 were purchased through the Schmooze portal. Should I contact Schmooze sales support?

I have no idea if it’s a codec issue–doesn’t seem like it should be since ulaw works fine if it’s the phone that initiates the page. In other words; when I use the phone’s web GUI to set one of the programmable buttons to “Multicast Paging” mode and set a multicast address and port the audio sounds perfect. And by default, the 2130 uses PCMU multicast.

When I use FreePBX/Asterisk to send the multicast page the audio sounds like it’s under water.

Thanks – Steve


(Tony Lewis) #19

You can open a support ticket under commercial modules at support.schmoozecom.com and we can try and setup a 2160 with asterisk and do multicast but that is about the extent of support we could give with the purchase of the commercial module as the module itself doesnt do much but tell Asterisk to start a multicast page.

Can you try a Multicast page to 1 or 2 phones only and see what happens. I wonder if its a codec issue but would be interested to know if paging a 1 or 2 phones at a time has the same issue.


(Steve) #20

I’m only using 2 phones at the moment and the problem persists. The most number of phones I’ve tried this with is 3 so it doesn’t appear to be related to a large number of phones.


(Steve) #21

I’ve edited the file and have tried alaw and g726 and the results are pretty much the same. Not sure if I needed to but I restarted asterisk after each change. Also confirmed that it was using the new codec by running the ‘core show channel Multi’ command.

I opened a support ticket open with Grandstream last Friday AM and so far response has been VERY slow; one response on Monday asking if the phones were being powered by PoE or via the external PS that ships with the phone. I responded immediately and nothing since.

I’m opening a support ticket with Schmooze now if for no other reason than to try to confirm they’re seeing the same problem.


#22

I am butting my head on this issue now in May 2017 and cannot find out what happened with it as far as resolution. No followup posted here. GXP2135 1.0.7.99

Is there a difference in packet size (payload) specified by assigning MPK button on Grandstream phone for multicast page, as opposed to the PagePro module? In a snom forum this issue was mentioned as a possible problem.

One config is generated by the phone’s firmware and the other by the PP module, right?

Sip Sip


(Tony Lewis) #23

Sorry but the issue here is not a FreePBX issue. This all handled inside Asterisk. We have no control over the Multicast page that asterisk sends out.