[asterisk-users] Grandstream GXP2140

2015-04-14 Thread Patrick Beaumont 
Hi Everyone.


I have a customer looking to deploy about 20 Grandstream GXP2140 phones. 
Normally they would deploy Yealink brand phones but they want a phone with 
gigabit pass through and the Yealinks with gigabit are too expensive for their 
budget.


Does anyone on the list have experience with the GXP2140? Is it a reliable 
phone? Does anyone have recommendations for other phones with gigabit pass 
through?


Regards,

Patrick.
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Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

2015-04-08 Thread Patrick Beaumont 
I have seen a similar problem occasionally. We will be doing maintenance on a 
customer's server and they will have one or two "ghost" channels on their 
machine hundreds of hours old but with no call associated with them. So far we 
have just been rebooting their server or issuing a hangup command to the 
channels.

From: asterisk-users-boun...@lists.digium.com 
 on behalf of Alex Villací­s Lasso 

Sent: 08 April 2015 00:33
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Help debugging a possible SIP channel leak in 
11.17.0, possible race condition

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:
> I am trying to collect enough information about an problem a client is having 
> with its asterisk 11.17.0  x86_64. This issue was observed before in 1.8.20, 
> and we upgraded to 11.15.0 and then to 11.17.0 with no solution.
>
> Background: this client is a telemarketing call-center that generates 
> outgoing calls with close to a hundred agents operating simultaneously in 
> peak hours. The system uses asterisk with FreePBX 2.8. In order to generate 
> the calls, I wrote a program that
> connects to Asterisk using the AMI protocol. This program expects the SIP 
> agent extensions to be assigned as members of queues, of which there are 
> about 20, as shown below:
>
> 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s 
> talktime), W:0, C:581, A:260, SL:82.6% within 60s
>Members:
>   SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls 
> (last was 800 secs ago)
>   SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls 
> (last was 708 secs ago)
>   SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls 
> (last was 656 secs ago)
>   SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls 
> (last was 789 secs ago)
>No Callers
>
> The program runs "queue show" through AMI every few seconds. For each queue 
> to be used in telemarketing, the program counts the number of members that 
> are "Not In Use". If at least one is found, it reads that many phone numbers 
> from the database and uses
> the AMI Originate command on each one, as follows:
>
> Action: Originate
> Channel: Local/NN@from-internal
> Exten: 
> Context: from-internal
> Priority: 1
> Async: true
> ActionID: xxx
>
> Here, NN is the number read from the database and  is the queue 
> extension in the FreePBX-created context that eventually runs the Queue() 
> dialplan application for the corresponding queue. This causes the call to be 
> connected between the
> outgoing number and the queue, and is then assigned to a queue member by 
> Asterisk. The dialplan is configured to route NN through one of a 
> series of SIP trunks using the outbound routes as configured by FreePBX.
>
> The issue is that although this strategy works correctly on the user's 
> machine for a few days, we have been observing that eventually the 
> application stops placing calls. The agents are all idle (all 90 to 100 of 
> them), but the "queue show" command shows
> them to be "In Use" on all queues. Furthermore, in normal operation, the 
> "core show channels" command shows at most one channel for each configured 
> SIP client in the "Up" state, but when calls stop being placed, the same 
> command reports multiple channels
> in the "Up" state, as follows (after sorting):
>
> Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192
> Local/9759315789@from-internal-a456;1!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
>  Line)!9759315789!!!3!500!(None)!1428426084.169326
> Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323
> SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224
> SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
> Line)!110!!!3!590!(None)!1428425994.169124
> SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
> SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273
> SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
> SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850
> SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113
> SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
> SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47

Re: [asterisk-users] Call Quality Measuring

2015-03-31 Thread Patrick Beaumont 
Thanks for the suggestions guys. I’ll try to have a play with Voipmonitor
in the near future.

So can I assume from the lack of discussion nobody is using the “sip show
channelstats” stuff?

Regards,
Patrick.

On 31/03/2015 08:23, "Olivier"  wrote:

>Some SIP hardphones (Polycom) or softphones (Counterpath) embed a
>module that metter MOS.
>
>
>Regards
>
>2015-03-25 14:21 GMT+01:00 Patrick Beaumont
>:
>> Hi everyone.
>>
>> We regularly get customers complaining about call quality issues. Most
>>of
>> the time it turns out to be their own broadband. Very occasionally
>>server
>> load. Does anyone have any advice or links to advice on measuring call
>> quality?
>>
>> I’ve been playing around with “sip show channelstats” but can’t other
>>than
>> measuring the packet loss I don’t really know what I’m supposed to be
>> looking for in order to say “ah ha! that’s the problem!”. I also don’t
>> know what it’s limits are. Will the stats in “sip show channelstats”
>>show
>> a customer using a torrent client and saturating their own broadband
>> connection?
>>
>> Regards,
>> Patrick.
>>
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Call Quality Measuring

2015-03-25 Thread Patrick Beaumont 
Hi everyone.

We regularly get customers complaining about call quality issues. Most of
the time it turns out to be their own broadband. Very occasionally server
load. Does anyone have any advice or links to advice on measuring call
quality?

I’ve been playing around with “sip show channelstats” but can’t other than
measuring the packet loss I don’t really know what I’m supposed to be
looking for in order to say “ah ha! that’s the problem!”. I also don’t
know what it’s limits are. Will the stats in “sip show channelstats” show
a customer using a torrent client and saturating their own broadband
connection?

Regards,
Patrick.

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Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration [Spam score:11%]

2015-01-23 Thread Patrick Beaumont 
I encountered a bug in some Polycom models where it would refuse to register to 
a domain that started with a number (e.g 3something.voip.com). Could that be 
applicable here?

Regards,
Patrick.

From: Jordan Cook - Gyron Networks 
mailto:jordan.c...@gyron.net>>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Date: Friday, 23 January 2015 16:24
To: "asterisk-users@lists.digium.com" 
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] Polycom SoundStation 6000 Dropping Registration [Spam 
score:11%]

Hello,

I'm having a problem with a few Polycom SoundStation 6000s. Everything works 
fine, but they drop registration to asterisk after about maybe 30 minutes – the 
phone does not re-try to register and if you try to dial out on the phone it 
says “URI Dialing is Disabled”

Has anyone else had this issue? I'm running asterisk 11.7.0.


This message may be private and confidential. If you have received this message 
in error, please notify us and remove it from your system.

Gyron may monitor email traffic data and the content of email for the purposes 
of security and staff training.

Gyron Internet Ltd is a limited company registered in England and Wales. 
Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel 
Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark.

Gyron is a Deloitte Technology Fast 50 ranked company.
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Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue [Spam score:10%]

2015-01-09 Thread Patrick Beaumont 
My suspicion would be that the line

o=Z 0 0 IN IP4 146.115.163.234​


is causing the problem. Your SIP client is reporting it's external IP address 
for the audio stream rather than it's internal one. I would look at the 
settings in your sip client to see if it has any automatic NAT stuff (like 
using a STUN server) and disable it.


Regards,

Patrick.


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Sonny Rajagopalan 

Sent: 09 January 2015 01:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue [Spam 
score:10%]

Well, I thought it worked, but it actually doesn't--I am able to get the caller 
pick up the phone, but for some reason, I cannot hear anything on either side 
no matter who does the calling. Again, my two SIP phones are on the local 
192.168.1.0/24 network (do not go over the Internet) and 
the Asterisk server is located in the same network (not accessed over the 
Internet). Any help is appreciated.

Does the fact that Asterisk is running on a VirtualBox VM on the same machine 
as one of the SIP phones matter? I am able to access the ARI REST interface of 
the Asterisk server quite fine on the host machine.

I suspect it has to do with RTP not being set up, but all the codec support is 
there. Here's a log for the SIP request from 192.168.1.50:

<--- Received SIP request (1229 bytes) from 
UDP:192.168.1.50:64009 --->
INVITE sip:6002@192.168.1.139;transport=UDP 
SIP/2.0
Via: SIP/2.0/UDP 
146.115.163.234:64009;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Max-Forwards: 70
Contact: 
To: mailto:sip%3A6002@192.168.1.139>;transport=UDP>
From: 
mailto:sip%3Ademo-alice@192.168.1.139>;transport=UDP>;tag=b661670b
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, 
SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.21933 r21903

Authorization: Digest 
username="demo-alice",realm="asterisk",nonce="[removed]",uri="sip:6002@192.168.1.139;transport=UDP",response="[removed]",cnonce="[removed]",nc=0001,qop=auth,algorithm=md5,opaque="[removed]"

Allow-Events: presence, kpml
Content-Length: 245


v=0
o=Z 0 0 IN IP4 146.115.163.234
s=Z
c=IN IP4 146.115.163.234
t=0 0
m=audio 8000 RTP/AVP 0 3 110 8 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


<--- Transmitting SIP response (319 bytes) to 
UDP:192.168.1.50:64009 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
146.115.163.234:64009;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
From: 
mailto:sip%3Ademo-alice@192.168.1.139>>;tag=b661670b
To: mailto:sip%3A6002@192.168.1.139>>
CSeq: 2 INVITE
Content-Length:  0

Any help is appreciated. A topology is shown below in ASCII.


  < ( Big bad Internet ) >

 GW/NAPT/Router
|
  --
 /   \
||
   Host A   Host B
-   
-
| Alice |   | Bob   
|
| 192.168.1.50  |   | 192.168.1.149 
|
|---|   
|---|
| Asterisk sr   |
|(VM)   |
| 192.168.1.239 |
|---|

On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan 
mailto:sonny.rajagopa...@gmail.com>> wrote:
Thank you for your note, Scott.

I set rewrite_contact=yes for both contacts, and I also had to do 
remove_existing=yes because I had to remove the existing contact information 
(max_contacts = 1 was preventing new contact information) using pjsip qualify 
demo-alice etc., after which the right IP addresses showed in pjsip show 
endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.1.0/24
;Templates for the necessary config sections

[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=ulaw

[auth_userpass](!)
type=auth
auth_type=userpass

[aor_dynamic](!)
type=aor
max_contacts=1
remove_existing=yes
;Definitions for our phones, using the templates above

[demo-alice](endpoint_internal)
auth=demo-alice
aors=demo-alice
mailboxes=box_a
rewrite_co

Re: [asterisk-users] status - Unmonitored, how to change it [Spam score:11%]

2014-12-30 Thread Patrick Beaumont 
I believe the "Unmonitored" status is linked to the "qualify" setting for each 
user. If they aren't set to "qualify=yes" then it won't check their status.

Regards,
Patrick.

From: asterisk-users-boun...@lists.digium.com 
 on behalf of Joseph 

Sent: 30 December 2014 18:58
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] status - Unmonitored, how to change it [Spam 
score:11%]

How to change status of peers "Unmonitored" to monitored?

Home users showing "Unmonitored" some display timing.

Name/UsernameHost Mask Port  Status
zoiper_kathy/zo  112.200.83.69   (D)  255.255.255.255  9330  Unmonitored
clinic_server(null)  (D)  255.255.255.255  0 Unmonitored
voip 184.89.249.114  (S)  255.255.255.255  4569  OK (91 ms)


--
Joseph

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Re: [asterisk-users] 11.5.0: blindxfer problems [Spam score:10%]

2014-12-21 Thread Patrick Beaumont 
Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using “sip
info” for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
dialled any digits.




On 20/12/2014 20:52, "sean darcy"  wrote:

>On 12/20/2014 03:22 PM, sean darcy wrote:
>> On 12/19/2014 09:42 AM, Rusty Newton wrote:
>>> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy 
>>>wrote:
 I've got a confbridge set up which works if dialed locally:

 -- Executing [266@internal:1] Answer("DAHDI/1-1", "") in new stack
  -- Executing [266@internal:2] SendDTMF("DAHDI/1-1", "1") in new
 stack
  -- Executing [266@internal:3] ConfBridge("DAHDI/1-1", "1") in
 new stack
  --  Playing 'conf-onlyperson.ulaw' (language 'en')
 ...


 extensions.conf:

 [globals]
 ...
 GOTO_ON_BLINDXFR="internal,266,1"

 features.conf:

 [featuremap]
 blindxfer => #1

 But:

 -- Executing [s@DialOut:14] Dial("DAHDI/1-1",
 "motif//+12345678...@voice.google.com,,rTt") in new stack
  -- Called motif//+12345678...@voice.google.com
  -- Motif/+12345678...@voice.google.com-688c is proceeding
 passing it to
 DAHDI/1-1
  -- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
  -- Started music on hold, class 'default', on
 Motif/+1234567...@voice.google.com-688c
  --  Playing 'pbx-transfer.ulaw' (language 'en')
 [Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550
 builtin_blindtransfer: No digits dialed.
  --  Playing 'pbx-invalid.ulaw' (language 'en')

 I'm expecting the blind transfer to GOTO internal,266,1.

 If I input 266 at the transfer dial tone, the blind transfer occurs.

 Do I have this set up incorrectly?
>>>
>>> 
>>>https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Var
>>>iables
>>>
>>>
>>> "${GOTO_ON_BLINDXFR} - Transfer to the specified
>>> context/extension/priority after a blind transfer (use ^ characters in
>>> place of | to separate context/extension/priority when setting this
>>> variable from the dialplan)"
>>>
>>> Try using ^ characters as it mentions there.
>>>
>>
>> Thanks for the response, but no joy:
>>
>>
>>   == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1'
>>
>>  Playing 'pbx-transfer.ulaw' (language 'en')
>> [Dec 20 15:12:03] WARNING[12336][C-0012]: features.c:2550
>> builtin_blindtransfer: No digits dialed.
>>
>>
>> sean
>>
>>
>
>I also tried setting up a transfer as an applicationmap.
>
>conference => *7,peer/both,ConfBridge,1
>
>Seems to load:
>
>features reload
>   == Parsing '/etc/asterisk/features.conf': Found
>   == Registered Feature 'conference'
>   == Mapping Feature 'conference' to app 'ConfBridge(1)' with code '*7'
>
>but when the caller dials *7, there's no action, Nothing in the cli. The
>dtmf is just sent to the callee.
>
>Also tried having the callee dial *7, same result.
>
>Any help appreciated.
>
>
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Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] [Spam score:8%]

2014-12-10 Thread Patrick Beaumont 
Thank you once again Richard. I think that covers all my confusion.

Regards,
Patrick.

From: Richard Mudgett mailto:rmudg...@digium.com>>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Date: Tuesday, 9 December 2014 21:48
To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam 
score:8%] [Spam score:8%]



On Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont 
mailto:p.beaum...@hatsoffsoftware.co.uk>> 
wrote:

Thanks Richard. This is exactly the answer I was looking for.


I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but 
I was getting confused because the only bridge module I saw in modules.conf was 
bridge_softmix. When I upgraded to Asterisk13 that would have been the only 
bridge getting loaded at first.


Is it expected that if bridge_softmix handled a normal two party call then MOH 
would no longer function?

That is correct.  bridge_softmix is optimized for multi-party conferencing 
where passing
control frames such as hold/unhold to other parties in the bridge is not a good 
idea.  For
example, if three parties are in a bridge and if party A pressed its hold 
button then that
should not necessarily prevent parties B and C from talking to each other.  
Using
bridge_softmix for a normal two party call is a last resort.  It works 
reasonably well as a
normal two party bridge technology but it is computationally expensive and not 
intended
for that purpose.

Richard

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Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%]

2014-12-09 Thread Patrick Beaumont 
Thanks Richard. This is exactly the answer I was looking for.


I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but 
I was getting confused because the only bridge module I saw in modules.conf was 
bridge_softmix. When I upgraded to Asterisk13 that would have been the only 
bridge getting loaded at first.


Is it expected that if bridge_softmix handled a normal two party call then MOH 
would no longer function?



From: asterisk-users-boun...@lists.digium.com 
 on behalf of Richard Mudgett 

Sent: 09 December 2014 20:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam 
score:8%]



On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont 
mailto:p.beaum...@hatsoffsoftware.co.uk>> 
wrote:

Hi Everyone.


I was referred here by malcolmd of the Asterisk forums. What follows is a copy 
of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007?


I've recently upgraded from Asterisk 11 to Asterisk 13.

Most of it went smoothly thanks to the documentation detailing how to upgrade 
to 12 and then how to upgrade to 13.

The only thing that didn't work correctly was Music On Hold. Eventually I 
tracked this down to using bridge_softmix instead of bridge_simple.

What I'm asking is, does anyone have any explanation as to why MOH would not 
work with bridge_softmix? Asterisk 11 had been working for at least a year with 
bridge_softmix and the MOH was fine. With the same configuration (almost) 
Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI 
about hold music starting or stopping. Unloading bridge_softmix and then 
loading bridge_simple fixes the issue.

Also does anyone have any documentation on what bridges I should be using? I 
can't seem to find anything in the upgrade documentation that says "MOH will no 
longer work in softmix, you should use simple". This has me concerned that I've 
done something wrong elsewhere in my config that is causing softmix to not work 
correctly.?

The bridging technology bridge_softmix is only used by app_confbridge in 
Asterisk v11.
Nothing else in v11 uses the bridging framework.  Unless you were using 
app_confbridge,
you were not using bridge_softmix in v11.

The various bridging technology modules in v12 and later are for different 
scenarios.  The
bridging framework is smart enough to pick the best bridging technology 
available for the
situation.  If the situation changes during a call, the bridging framework can 
change the
bridge technology to support the new situation.

* bridge_simple is for normal two party communication.

* bridge_native_rtp is a special case of two party bridge were both parties use 
RTP for
media exchange.  The native technology allows for direct media.

* bridge_softmix is for multi-party bridges where you can have 1 to n users 
communicating
in a conference.  As you found out, bridge_softmix can be used as a fallback if 
bridge_simple
is not available because it allows two party communication.

* bridge_holding is a parking bridge technology to hold calls for later 
connection.  Parties
in a holding bridge cannot communicate with each other.

* bridge_builtin_features and bridge_builtin_interval_feature provide 
functionality used by
features.conf.  These two modules are actually not bridging technologies but 
support code
for features.conf functionality.

You usually need to install all of the bridging technologies.

Richard

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[asterisk-users] Bridge configuration in Asterisk 13

2014-12-09 Thread Patrick Beaumont 
Hi Everyone.


I was referred here by malcolmd of the Asterisk forums. What follows is a copy 
of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007?


I've recently upgraded from Asterisk 11 to Asterisk 13.

Most of it went smoothly thanks to the documentation detailing how to upgrade 
to 12 and then how to upgrade to 13.

The only thing that didn't work correctly was Music On Hold. Eventually I 
tracked this down to using bridge_softmix instead of bridge_simple.

What I'm asking is, does anyone have any explanation as to why MOH would not 
work with bridge_softmix? Asterisk 11 had been working for at least a year with 
bridge_softmix and the MOH was fine. With the same configuration (almost) 
Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI 
about hold music starting or stopping. Unloading bridge_softmix and then 
loading bridge_simple fixes the issue.

Also does anyone have any documentation on what bridges I should be using? I 
can't seem to find anything in the upgrade documentation that says "MOH will no 
longer work in softmix, you should use simple". This has me concerned that I've 
done something wrong elsewhere in my config that is causing softmix to not work 
correctly.?


Regards,
Patrick.

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