[asterisk-users] Choppy MOH (Cisco gateway)

John Sawa john.sawa at oracle.com
Sun Jul 9 19:52:25 MST 2006


You will also want to add

no vad 

to your dial-peer config to disable voice activity detection.

I do not think it will resolve your issue, but worth a shot.

-John

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of 
> Bill Gibbs
> Sent: Sunday, July 09, 2006 7:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway)
> 
> 
> 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
> > 
> > 
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 




More information about the asterisk-users mailing list