Re: [asterisk-users] They ignore my DTMF!
Benjamin Jacob wrote: rfc2833 is the prefered way, as inband will work perfectly only with the ulaw codec. Out of curiosity, there is any 'document' about how RFC2833 would be better or worse than SIP INFO ? Pierre Marceau wrote: Okay, in the SPA-941 admin I changed: ;DTMF Tx Method: Auto DTMF Tx Method: Inband and now it works. Thanks! Pierre [EMAIL PROTECTED] 2/21/2007 12:09 AM Pierre, Thats exactly what Joanna said in her reply. Check the client DTMF settings on your phones. set it to rfc2833 or out-of-band, whatever the config says. Grandstream by default have inband DTMF set, and usualy ulaw is supported as well, and thats the reason ur grandstream works but others dont. cheerz - Ben. Pierre Marceau wrote: Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. ___ --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
[asterisk-users] They ignore my DTMF!
Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --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
Re: [asterisk-users] They ignore my DTMF!
Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --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
Re: [asterisk-users] They ignore my DTMF!
Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --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
Re: [asterisk-users] They ignore my DTMF!
Pierre, Thats exactly what Joanna said in her reply. Check the client DTMF settings on your phones. set it to rfc2833 or out-of-band, whatever the config says. Grandstream by default have inband DTMF set, and usualy ulaw is supported as well, and thats the reason ur grandstream works but others dont. cheerz - Ben. Pierre Marceau wrote: Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --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 -- The problem with the Future is that it keeps turning into the Present. ___ --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
Re: [asterisk-users] They ignore my DTMF!
Okay, in the SPA-941 admin I changed: ;DTMF Tx Method: Auto DTMF Tx Method: Inband and now it works. Thanks! Pierre [EMAIL PROTECTED] 2/21/2007 12:09 AM Pierre, Thats exactly what Joanna said in her reply. Check the client DTMF settings on your phones. set it to rfc2833 or out-of-band, whatever the config says. Grandstream by default have inband DTMF set, and usualy ulaw is supported as well, and thats the reason ur grandstream works but others dont. cheerz - Ben. Pierre Marceau wrote: Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --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 -- The problem with the Future is that it keeps turning into the Present. ___ --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
Re: [asterisk-users] They ignore my DTMF!
rfc2833 is the prefered way, as inband will work perfectly only with the ulaw codec. Pierre Marceau wrote: Okay, in the SPA-941 admin I changed: ;DTMF Tx Method: Auto DTMF Tx Method: Inband and now it works. Thanks! Pierre [EMAIL PROTECTED] 2/21/2007 12:09 AM Pierre, Thats exactly what Joanna said in her reply. Check the client DTMF settings on your phones. set it to rfc2833 or out-of-band, whatever the config says. Grandstream by default have inband DTMF set, and usualy ulaw is supported as well, and thats the reason ur grandstream works but others dont. cheerz - Ben. Pierre Marceau wrote: Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --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 -- The problem with the Future is that it keeps turning into the Present. ___ --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
Re: [asterisk-users] They ignore my DTMF!
Hi Pierre, You can also add the following if you think its helpful. relaxdtmf = yes ;relaxes the DTMF detection parameters qualify = yes ; will send a SIP Optoin command regularly to check if the device is still online, if the device did not answer within 2 seconds it will be considered offline (default time is 2 seconds but can be configured) Best Regards, Joanna On 2/21/07, Benjamin Jacob [EMAIL PROTECTED] wrote: rfc2833 is the prefered way, as inband will work perfectly only with the ulaw codec. Pierre Marceau wrote: Okay, in the SPA-941 admin I changed: ;DTMF Tx Method: Auto DTMF Tx Method: Inband and now it works. Thanks! Pierre [EMAIL PROTECTED] 2/21/2007 12:09 AM Pierre, Thats exactly what Joanna said in her reply. Check the client DTMF settings on your phones. set it to rfc2833 or out-of-band, whatever the config says. Grandstream by default have inband DTMF set, and usualy ulaw is supported as well, and thats the reason ur grandstream works but others dont. cheerz - Ben. Pierre Marceau wrote: Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --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 -- The problem with the Future is that it keeps turning into the Present. ___ --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
