Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN through the FXO cards I get horrible echo, I have even been able when talking loud enough to get a horrible feedback loop going. I have tried 4 different echo cancellers in the Makefile for the Zap drivers and nonoe of them changed the situation. I have echocancel = (Any where from 1 - 256, I have tried alot of different values), and I have echocanelwhenbridged = yes.I only hear the echo start when the call gets bridged onto the outgoing PSTN lines. Is there anything I can do? Brian J. Schrock
----- Original Message ----- From: "Brian J. Schrock" <brians@anistonetech.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, August 28, 2003 6:16 PM Subject: [Asterisk-Users] SIP and ECHO> Hello, > > I have read the information on echo and SIP in the FAQ and I have > scoured the mailing list for possible solutions, but as yet I have not > been able to get rid of this echo. > > I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed > into an asterisk server. If I call between the Sip Phone > (Budgettone-100) and the 4 FXS ports everything sounds great. If I call > out to the PSTN through the FXO cards I get horrible echo, I have even > been able when talking loud enough to get a horrible feedback loop > going. I have tried 4 different echo cancellers in the Makefile for the > Zap drivers and nonoe of them changed the situation. > > I have echocancel = (Any where from 1 - 256, I have tried alot of > different values), and I have echocanelwhenbridged = yes.I only hear the > echo start when the call gets bridged onto the outgoing PSTN lines. > > Is there anything I can do? > > Brian J. Schrock >Hi, For me: rxgain=0.8 txgain=0.8 in zapata conf do the trick. Now the echo is allmost inexistant. Maybe the sound is not very strong but the quality is very good. I have the default echo canceller (no modification in the source files). Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711), Cisco 79x0) and one X100P card. BR, Dan
I can minimize doing those tricks, but I cannot seem to get it to go away. On Thu, 2003-08-28 at 11:33, Dan wrote:> ----- Original Message ----- > From: "Brian J. Schrock" <brians@anistonetech.com> > To: <asterisk-users@lists.digium.com> > Sent: Thursday, August 28, 2003 6:16 PM > Subject: [Asterisk-Users] SIP and ECHO > > > > Hello, > > > > I have read the information on echo and SIP in the FAQ and I have > > scoured the mailing list for possible solutions, but as yet I have not > > been able to get rid of this echo. > > > > I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed > > into an asterisk server. If I call between the Sip Phone > > (Budgettone-100) and the 4 FXS ports everything sounds great. If I call > > out to the PSTN through the FXO cards I get horrible echo, I have even > > been able when talking loud enough to get a horrible feedback loop > > going. I have tried 4 different echo cancellers in the Makefile for the > > Zap drivers and nonoe of them changed the situation. > > > > I have echocancel = (Any where from 1 - 256, I have tried alot of > > different values), and I have echocanelwhenbridged = yes.I only hear the > > echo start when the call gets bridged onto the outgoing PSTN lines. > > > > Is there anything I can do? > > > > Brian J. Schrock > > > > > Hi, > > For me: > > rxgain=0.8 > txgain=0.8 > > in zapata conf do the trick. > Now the echo is allmost inexistant. Maybe the sound is not very strong but > the quality is very good. > I have the default echo canceller (no modification in the source files). > > Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711), > Cisco 79x0) and one X100P card. > > BR, > Dan > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Ever since updating through cvs I have also noticed echo on the pstn lines using a fxo channel bank. I had no echo prior to the cvs update. I have backed off the gain for the FXO channels also (from 2.0 to 0.0) and this helped but there is still sometimes (but not all the time) a very distinct echo. It is like sometimes the echo canceling works and sometimes it does not. I am now running asterisk cvs-08/26/03-14:04:16. I also did a cvs update for Zaptel and libpri on August 26th since * would not compile without it. Previously I was running the cvs version that I downloaded on April 16, 2003. I am not using a sip phone but an ADSI analog set so it appears that this is not sip related. If someone could verify this, a bug could be filed in bugtracker. Don Pobanz On Thursday, August 28, 2003 10:34 AM, Dan [SMTP:dtoma@fx.ro] wrote:> > ----- Original Message ----- > From: "Brian J. Schrock" <brians@anistonetech.com> > To: <asterisk-users@lists.digium.com> > Sent: Thursday, August 28, 2003 6:16 PM > Subject: [Asterisk-Users] SIP and ECHO > > > > Hello, > > > > I have read the information on echo and SIP in the FAQ and I have > > scoured the mailing list for possible solutions, but as yet I have > > not > > been able to get rid of this echo. > > > > I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards > > installed > > into an asterisk server. If I call between the Sip Phone > > (Budgettone-100) and the 4 FXS ports everything sounds great. If I > > call > > out to the PSTN through the FXO cards I get horrible echo, I have > > even > > been able when talking loud enough to get a horrible feedback loop > > going. I have tried 4 different echo cancellers in the Makefile for > > the > > Zap drivers and nonoe of them changed the situation. > > > > I have echocancel = (Any where from 1 - 256, I have tried alot of > > different values), and I have echocanelwhenbridged = yes.I onlyhear> > the > > echo start when the call gets bridged onto the outgoing PSTN lines. > > > > Is there anything I can do? > > > > Brian J. Schrock > > > > > Hi, > > For me: > > rxgain=0.8 > txgain=0.8 > > in zapata conf do the trick. > Now the echo is allmost inexistant. Maybe the sound is not verystrong> but > the quality is very good. > I have the default echo canceller (no modification in the source > files). > > Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone > (G.711), > Cisco 79x0) and one X100P card. > > BR, > Dan
I've been having the same problem too, except for me it only happens occasionnally. I'm not 100% sure of this, but it seems that for very local calls (eg across the city) I never get echo. For calls that go longer distance (say 500km or more), or through some closer call centres, I'm getting the echo. I don't get the echo on an analogue POTS connection to the same places (it is clearly only happening on our asterisk system). This might indicate some link between echo cancellation and delayed audio, but if so, its sensitive to very small delays. The echo can only be heard at our end, there is no trace of it at the other end. I'm using ATAs doing SIP to Asterisk and through a PRI connection to a Telco. Echo cancellation is turned on and showing as activated on the Zap channels. Echo cancellation is also enabled on the ATAs. ----- Original Message ----- From: "Brian J. Schrock" <brians@anistonetech.com> To: <asterisk-users@lists.digium.com> Sent: Friday, August 29, 2003 3:16 AM Subject: [Asterisk-Users] SIP and ECHO> Hello, > > I have read the information on echo and SIP in the FAQ and I have > scoured the mailing list for possible solutions, but as yet I have not > been able to get rid of this echo. > > I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed > into an asterisk server. If I call between the Sip Phone > (Budgettone-100) and the 4 FXS ports everything sounds great. If I call > out to the PSTN through the FXO cards I get horrible echo, I have even > been able when talking loud enough to get a horrible feedback loop > going. I have tried 4 different echo cancellers in the Makefile for the > Zap drivers and nonoe of them changed the situation. > > I have echocancel = (Any where from 1 - 256, I have tried alot of > different values), and I have echocanelwhenbridged = yes.I only hear the > echo start when the call gets bridged onto the outgoing PSTN lines. > > Is there anything I can do? > > Brian J. Schrock > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I've been having the same problem too, except for me it only happens occasionnally. I'm not 100% sure of this, but it seems that for very local calls (eg across the city) I never get echo. For calls that go longer distance (say 500km or more), or through some closer call centres, I'm getting the echo. I don't get the echo on an analogue POTS connection to the same places (it is clearly only happening on our asterisk system). This might indicate some link between echo cancellation and delayed audio, but if so, its sensitive to very small delays. The echo can only be heard at our end, there is no trace of it at the other end. I'm using ATAs doing SIP to Asterisk and through a PRI connection to a Telco. Echo cancellation is turned on and showing as activated on the Zap channels. Echo cancellation is also enabled on the ATAs. ----- Original Message ----- From: "Brian J. Schrock" <brians@anistonetech.com> To: <asterisk-users@lists.digium.com> Sent: Friday, August 29, 2003 3:16 AM Subject: [Asterisk-Users] SIP and ECHO> Hello, > > I have read the information on echo and SIP in the FAQ and I have > scoured the mailing list for possible solutions, but as yet I have not > been able to get rid of this echo. > > I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed > into an asterisk server. If I call between the Sip Phone > (Budgettone-100) and the 4 FXS ports everything sounds great. If I call > out to the PSTN through the FXO cards I get horrible echo, I have even > been able when talking loud enough to get a horrible feedback loop > going. I have tried 4 different echo cancellers in the Makefile for the > Zap drivers and nonoe of them changed the situation. > > I have echocancel = (Any where from 1 - 256, I have tried alot of > different values), and I have echocanelwhenbridged = yes.I only hear the > echo start when the call gets bridged onto the outgoing PSTN lines. > > Is there anything I can do? > > Brian J. Schrock > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
chan_h323 is built into asterisk. Check the /usr/src/asterisk/channels/h323 directory for more info. ----- Original Message ----- From: "Phillip Britt" <phil@wideband.net.au> To: <asterisk-users@lists.digium.com> Sent: Tuesday, September 02, 2003 1:12 PM Subject: [Asterisk-Users] H.323 Support Hi, I am currently using Asterisk and want to add H.323 support for talking to our gateway routers, which use gnkgk Is the package "Asterisk-oh323" the right thing to use, or are there better ways of achieving h.323 support in Asterisk. Thanks, Phil _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users