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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060709/2023df7f/attachment.htm
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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@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
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
Actually this seems to have fixed it!! Bill -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Sawa Sent: Sunday, July 09, 2006 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) 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@lists.digium.com > [mailto:asterisk-users-bounces@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@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mike > Sent: Monday, July 10, 2006 2:51 AM > To: asterisk-users@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 >_______________________________________________ --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
And of course I just found this article http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Hope this helps some other people out as well! Bill -----Original Message----- From: Bill Gibbs Sent: Monday, July 10, 2006 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) Actually this seems to have fixed it!! Bill -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Sawa Sent: Sunday, July 09, 2006 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) 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@lists.digium.com > [mailto:asterisk-users-bounces@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@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mike > Sent: Monday, July 10, 2006 2:51 AM > To: asterisk-users@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 >_______________________________________________ --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
On Jul 10, 2006, at 4:49 AM, Bill Gibbs wrote:> And of course I just found this article > > http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 > > Hope this helps some other people out as well! >So was the fix to reconfigure your gateway to not?use VAD? Just want to be clear... Marty
Yes that is correct. Bill -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Martin Joseph Sent: Monday, July 10, 2006 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) On Jul 10, 2006, at 4:49 AM, Bill Gibbs wrote:> And of course I just found this article > > http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 > > Hope this helps some other people out as well! >So was the fix to reconfigure your gateway to not?use VAD? Just want to be clear... Marty _______________________________________________ --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