Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. The TDMx00P cards are FXS cards.. :) 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. What are you using to connect to the PSTN?? X100P, T100P, E100P, I4L, Chan_Capi 2nd question: using a grandstream phone asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? You need to setup call groups, search through the archives cos I rememeber a thread on this a short while ago.. 3rd question: can someone give me some starter hints to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving person before actually passing the call. As far as I know there is no facility to do a consultative transfer on the GS phones.. Only a blind transfer.. Maybe it will come later.. can anybody help please ? cheers Dave A Caruana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
these are taken as db right? 3.0 db = 100% but, in some cases we've had to do txgain=9.0 is that bad, martin? are there any hardware limitations on this? does zaptel really accept %? if so, then it should be taken as a percentage, not pseudoDB (tm) - wasim On Tue, 5 Aug 2003, Martin Pycko wrote: Don't use %'s with txgain/rxgain for txgain=5% is equal to txgain=5.0 and that might be too much ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
my error .. the cards are X100P which is why I wrote FXO. The Grandstream phones are on a LAN, the * server connects to the phonelines via the X100P cards. When I call from the Grandstream phones onto the PSTN there is a VERY big amount of echo, ie. I can hear myself in the earpiece. cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 05, 2003 8:50 AM Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. The TDMx00P cards are FXS cards.. :) 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. What are you using to connect to the PSTN?? X100P, T100P, E100P, I4L, Chan_Capi 2nd question: using a grandstream phone asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? You need to setup call groups, search through the archives cos I rememeber a thread on this a short while ago.. 3rd question: can someone give me some starter hints to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving person before actually passing the call. As far as I know there is no facility to do a consultative transfer on the GS phones.. Only a blind transfer.. Maybe it will come later.. can anybody help please ? cheers Dave A Caruana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
could you send me the exact syntax for rxgain / txgain? I think that might help towards my problem becuase i'm having to turn the handset volume all the way up .. thanks Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 05, 2003 9:45 AM Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers my error .. the cards are X100P which is why I wrote FXO. The Grandstream phones are on a LAN, the * server connects to the phonelines via the X100P cards. When I call from the Grandstream phones onto the PSTN there is a VERY big amount of echo, ie. I can hear myself in the earpiece. cheers Dave An echo at the begining of a call is normal as the * and phone trains themselves but this should dissappear after about 30 seconds to 1 min.. So my only suggesttions are.. First make sure you have echocancel=yes and echocancelwhenbridged=yes in your zapata.conf.. If that doesn't help try lowering the volume on the sip handset and play with the rxgain= and txgain= in zapata.conf for the X100P's.. Other than that I don't really know what else you can try.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
I tried putting in txgain=100% rxgain=100% and zaptel wouldn't load telling me I had wrong parameters in my zaptel.conf i'll try again with txgain=5.0 but my setup is at a client so each time a day passes and i have to go round to the client just to try things out ... it's a bit annoying! my 2c .. when is there going to be some concerted effort at documenting some stuff? today I discovered by change that you can dial # to transfer to extension .. surely these are stuff that could be put down in writing somewhere ? cheers Dave - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 05, 2003 7:42 PM Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers Don't use %'s with txgain/rxgain for txgain=5% is equal to txgain=5.0 and that might be too much On Tue, 5 Aug 2003, WipeOut . wrote: could you send me the exact syntax for rxgain / txgain? I think that might help towards my problem becuase i'm having to turn the handset volume all the way up .. thanks Dave You can use either a percentage or a number IIRC.. Somthing like.. rxgain=5% txgain=5% or rxgain=0.4 txgain=0.4 and I thing that you can use negative values as well.. I am not sure what the minimum and maximum values are I use percntages.. Hope that helps.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
Dave Alan Caruana wrote: The Grandstream phones are on a LAN, the * server connects to the phonelines via the X100P cards. When I call from the Grandstream phones onto the PSTN there is a VERY big amount of echo, ie. I can hear myself in the earpiece. - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 05, 2003 8:50 AM Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. My configuration : 1 - X-TEL SIP phone - phone handset connected to sound blaster (providing no accoustic echo by itself. Tested with Netmeeting) 2 - one single FXO board (on the Asterisk side) 3 - remote = PSTN telephone set or GSM telephone set The symptoms : - broad local echo (IP side) - tiny echo on the PSTN side What I have tried : 1 - change the handset with an USB one (nothing change but the sound quality : worst !) 2 - change the echo canceller attached to the FXO board (nothing really noticiable) 3 - change to IAX (changing the client software) : seems to cancel the echo (!?) Problem stays alive ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
my error .. the cards are X100P which is why I wrote FXO. The Grandstream phones are on a LAN, the * server connects to the phonelines via the X100P cards. When I call from the Grandstream phones onto the PSTN there is a VERY big amount of echo, ie. I can hear myself in the earpiece. cheers Dave An echo at the begining of a call is normal as the * and phone trains themselves but this should dissappear after about 30 seconds to 1 min.. So my only suggesttions are.. First make sure you have echocancel=yes and echocancelwhenbridged=yes in your zapata.conf.. If that doesn't help try lowering the volume on the sip handset and play with the rxgain= and txgain= in zapata.conf for the X100P's.. Other than that I don't really know what else you can try.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
Don't use %'s with txgain/rxgain for txgain=5% is equal to txgain=5.0 and that might be too much On Tue, 5 Aug 2003, WipeOut . wrote: could you send me the exact syntax for rxgain / txgain? I think that might help towards my problem becuase i'm having to turn the handset volume all the way up .. thanks Dave You can use either a percentage or a number IIRC.. Somthing like.. rxgain=5% txgain=5% or rxgain=0.4 txgain=0.4 and I thing that you can use negative values as well.. I am not sure what the minimum and maximum values are I use percntages.. Hope that helps.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
