I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the Definity), and cell phones are great. When I call outside of the office to POTS lines (like my home), there is a most noticeable echo of my voice. The party on the line hears no echo. Any efforts on the configuring the SIP softphones, and within zapata.conf, have been for naught. Is this problem common for ISDN-PRI connections? _ This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050322/3e7d8dcf/attachment.htm
On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote:> I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an > Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover > cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of > the office calls go through the Definity. Here's the issue: > > Calls to internal SIP extensions, Definity extensions, other offices within > our private network (through the Definity), and cell phones are great. When > I call outside of the office to POTS lines (like my home), there is a most > noticeable echo of my voice. The party on the line hears no echo. Any > efforts on the configuring the SIP softphones, and within zapata.conf, have > been for naught. > > Is this problem common for ISDN-PRI connections?It is a problem you will see when calling an analogue subscriber over a link with a long latency (such as VoIP). The echo will most probably be generated by a 2- to 4-wire hybrid at the far end. In a pure amalogue/tdm path you would perceive the reflected energy as a plesent sidetone. As soon as the latency increases to 50-100ms the refelcted energy will be perceived as an echo instead. The options available to you are to live with the echo of your own voice or to insert an echo canceller at the pstn interface. Asterisk includes an echo canceller that may or may not be good enough. It seems to like some pstn interfaces and not others. If the Asterisk echo canceller is not enough you may consider an expensive inline echo canceller. Peter
>Peter Svensson wrote: > >>On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote: >> >> >> >>>I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an >>>Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover >>>cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of >>>the office calls go through the Definity. Here's the issue: >>> >>>Calls to internal SIP extensions, Definity extensions, other officeswithin>>>our private network (through the Definity), and cell phones are great.When>>>I call outside of the office to POTS lines (like my home), there is amost>>>noticeable echo of my voice. The party on the line hears no echo. Any >>>efforts on the configuring the SIP softphones, and within zapata.conf,have>>>been for naught. >>> >>>Is this problem common for ISDN-PRI connections? >>> >>> >> >>It is a problem you will see when calling an analogue subscriber over a >>link with a long latency (such as VoIP). The echo will most probably be >>generated by a 2- to 4-wire hybrid at the far end. In a pure amalogue/tdm >>path you would perceive the reflected energy as a plesent sidetone. As >>soon as the latency increases to 50-100ms the refelcted energy will be >>perceived as an echo instead. >> >>The options available to you are to live with the echo of your own voice >>or to insert an echo canceller at the pstn interface. Asterisk includes an>>echo canceller that may or may not be good enough. It seems to like some >>pstn interfaces and not others. If the Asterisk echo canceller is not >>enough you may consider an expensive inline echo canceller. >> >> > >The definity has echo cancellation. Try turning that on. > >-SteveKThanks for both of your replies. My Definity connection is on a TN464C (the TN464G has echo cancellation) and I was hoping to try a software solution first before investing in the hardware (we would also have to upgrade our Definity software, so the total cost is quite high). I have noticed that any of the zapata.conf echo cancel parameters seem to have no effect on an ISDN-PRI line, using pri_net signalling (I used the voip-info.org wiki for the configuration). If this is true, and I am not making some dumb mistake, is there another signalling mode that I can use on a Definity that will take advantage of echo-cancel? Barring that, I am using a Plantronics DSP-100 headset with X-lite under Windows. Can I set up echo-cancelling there? Thanks, Mark. P.S. Is there a website where I can post replies to this mailing list? Cutting and pasting into Outlook is just darn goofy. M. _ This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050323/4fb7e0bc/attachment.htm
> -----Original Message----- > From: McQuiggan, Mark xt46480 [mailto:Mark_McQuiggan@adp.com] > Sent: Wednesday, March 23, 2005 12:28 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] TE405P and echo > >Peter Svensson wrote: > >>On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote: >> >> >> >>>I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an >>>Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover >>>cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of >>>the office calls go through the Definity. Here's the issue: >>> >>>Calls to internal SIP extensions, Definity extensions, other offices within >>>our private network (through the Definity), and cell phones are great. When >>>I call outside of the office to POTS lines (like my home), there is a most >>>noticeable echo of my voice. The party on the line hears no echo. Any >>>efforts on the configuring the SIP softphones, and within zapata.conf, have >>>been for naught. >>> > {clip} > > I have noticed that any of the zapata.conf echo cancel parameters seem to have no effect on an ISDN-PRI line, using pri_net signalling (I used the voip-info.org wiki for the configuration). If this is true, and I am not making some dumb mistake, is there another signalling mode that I can use on a Definity that will take advantage of echo-cancel? >The echo cancellation code does not have any relationship to PRI-ness of at T1. So long as the channel is a B channel and the echo cancellation is correctly specified in zapata.conf then the code _will_ be active on the channel. This is the exact formatting I have used to specify bearer channels 1-23 of the PRI to my PBX (not a Definity): ; Norstar #1 (Wharf Road) context=in-t1nstar group=1 callerid=asreceived usecallerid=yes hidecallerid=no usecallingpres=yes rxgain=0 txgain=0 echocancel=yes echotraining=yes ;100 echocancelwhenbridged=no switchtype=dms100 pridialplan=local signalling=pri_net channel => 1-23 Then I can fire up the asterisk console and issue a 'zap show channel 10' and, when on hook (or its off hook and the echo can has been disabled by the relevant in band disable-tones) I see: ... Echo Cancellation: 128 taps unless TDM bridged, currently OFF ... If the channel is offhook I see: ... Echo Cancellation: 128 taps unless TDM bridged, currently ON ... If I were to specify a different number of taps for the echo can (ie. echocancel=256) then I would see the different number of taps in the 'zap show...' accordingly. If all of this agrees with your system then it follows that the code is indeed enabled and being called but the algorithm is failing for some reason. For that you would need to work on determining if the signal levels on the T1 are correct for the echo canceller implementation and adjust the relevant constants in zaptel/mec2_const.h if it's not. See http://bugs.digium.com/bug_view_page.php?bug_id=0002820 for a starting point. Also search the asterisk-users archives for a recent thread titled 'Tweaking AGGRESSIVE_SUPPRESSOR' Hope that helps. Kris Boutilier Information Services Coordinator Sunshine Coast Regional District
On Wed, 23 Mar 2005, McQuiggan, Mark xt46480 wrote:> I have noticed that any of the zapata.conf echo cancel parameters seem to > have no effect on an ISDN-PRI line, using pri_net signalling (I used the > voip-info.org wiki for the configuration). If this is true, and I am not > making some dumb mistake, is there another signalling mode that I can use on > a Definity that will take advantage of echo-cancel?The echo cancellation should be applied when you bridge a tdm channel to a non-tdm channel. To activate it even on a tdm bridge you have to enable the echocancelwhenbridged option. Peter