[asterisk-users] Choppy MOH (Cisco gateway)

[Bill Gibbs]

I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get
it.  I made sure to upgrade zaptel, etc as well.

I do have something of interest to note...
Placing the call on hold then taking it off hold and back on the music
is ok (doing that once it gets choppy) of course this is not practical
since the person using hold won't know if it's choppy.  It then gets
choppy again if you wait 15-20 secs.

I have 2 ways of making outbound calls from all of the boxes, and I did
the following via 1.2.9.1 and 1.2.4

1) Send the outbound call to the Cisco and send out via the PRI (sip
phone ulaw to Cisco ulaw out the PRI)
2) Dial "long distance" to a provider using g729 (Polycom to Asterisk
ulaw, Asterisk transcoding to g729 to provider)

If I call from a sip phone OUT to my cell via the long distance provider
I get no choppiness.   I am not able to get inbound calls from the
provider so I can only test one way.

So I then switched talking to my Cisco via g729 (letting asterisk
transcode ulaw to g729 and also g729 all the way through) and voice is
fine but MOH is still choppy.  So it must be something with the Cisco
maybe?  IOS version is 
Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6,
RELEASE SOFTWARE (fc2)

I have setup for the codecs:
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8

incoming dial-peer:

dial-peer voice 1 pots
 description Match all incoming calls, set DID
 incoming called-number .T
 direct-inward-dial
 forward-digits extra

dial-peer voice 16 voip
 description to the asterisk server
 destination-pattern <phone#>
 voice-class codec 1
 session protocol sipv2
 session target ipv4:<ip>
 dtmf-relay sip-notify rtp-nte

and outbound:

dial-peer voice 10000 pots
 description Outbound via PRI
 destination-pattern .T
 port 1/0:23
 forward-digits all

Could this have something to do with the Cisco suppressing the stream
using silence suppression...I read somewhere that Asterisk relies on Sip
packets for MOH??? 

There is not a bandwidth issue, the 3660 and boxes are on the same
switch VLAN w/ DSCP enabled.

Bill

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mike
Sent: Monday, July 10, 2006 2:51 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway)

i had a similar issue with the first branch of asterisk 1.2 and cheap
phones (tip-100 from tatung)
i'll suggest you to upgrade your asterisk box
are you using bristuff ?
try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1

lemme know
.mike


On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote:
> Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not
> connected, separate PBXs)  using ulaw all have issues with music on
> hold being choppy.  Normal voice and SIP (taking a call from the PRI,
> placing a call or extension to extension calls) conversations are
> _perfect_ with no drop outs so it's not a problem with the PRI or the
> 3660 talking to the Asterisk boxes.  If I call from my Polycom into an
> extension that immediately starts MusicOnHold it's perfect as well.
> 
>  
> 
> However, calling into the box via the PRI and being placed on hold the
> music is choppy.  Also, calling into an extension that spawns
> MusicOnHold immediately is choppy when it comes in via the Cisco.
> 
>  
> 
> This happens with mpg123, madplay and I tried using the Asterisk 1.2
> native mode in musiconhold.conf:
> 
>  
> 
> [default]
> 
> mode => files
> 
> directory => /var/lib/asterisk/mohmp3
> 
> random => yes
> 
>  
> 
> Same problem with all 3.
> 
>  
> 
> Tried converting MP3s to a pcm or ulaw file, same problem (using lame
> and sox to do the conversions)
> 
>  
> 
> It seems that this is common issue with no clear resolution.
> 
>  
> 
> Machines are Pentium 4s 512MB or 1GB RAM.  I would be the only call on
> the box, no load, etc.
> 
> Using ztdummy (or without, same behavior)
> 
> Asterisk ver 1.2.4 on all
> 
> Normal voice, IVR, play back voicemail, etc are all 100% perfect only
> on MusicOnHold has this issue
> 
> Polycom SIP phones or using X-Lite to test (used to make the call into
> MusicOnHold or answer the call coming in via the PRI and placing on
> hold)
> 
> Calling in from landline or cell phone - no difference
> 
>  
> 
> Any ideas?
> 
>  
> 
> Bill
> 
> 
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