[asterisk-users] Re: problems using the 1.4 version of meetme
The nearest I can do is I have a Linksys 3102 and its also set to inband and when I call that extension fromm outside using asterisk 1.4 I can hear the dtmf just fine in my ear -- works about the same as using 1.2. The only app so far which is not working is meetme not detecting the * in asterisk 1.4 using sip and this is what I need help with. Is this something I should file a bug about? on Friday 12/22/2006 Tony Mountifield([EMAIL PROTECTED]) wrote In article [EMAIL PROTECTED], John covici [EMAIL PROTECTED] wrote: The 1.4 and 1.2 are alternately on the same box -- I have to install 1.2 rmmod the zaptel modules delete /usr/lib/asterisk/modules/* and install the zaptel and asterisk. I use the same procedure to go back to the 1.4. I know the ivr works because I have to use a ivr menu and even enter a password to get to the meetme conference and those work fine. The sip provider is using inband as I have requested. Also, I tried calling through the sip provider to my local extension and I hear the tones just fine, so its a mystery to me. Do you have a SIP phone you can register directly with the box and try? If so, try setting it to the three different ways of sending DTMF and see whether any of them work. Just trying to whittle down the possibilities to start with You may find it better to use out-of-band DTMF with SIP. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] Re: problems using the 1.4 version of meetme
In article [EMAIL PROTECTED], John covici [EMAIL PROTECTED] wrote: Hi. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do the exacct same procedure but use a phone number which calls my box via a zap channel -- using a digium card -- it works perfectly. This problem seems to be independent of asterisk 1.4 and zaptel 1.4 versions, but I did an svn update this morning on both of those. Now this problem does NOT occurr with 1.2 at all, I can call my box using sip and the * is seen by the meetme conference. Are the 1.2 and 1.4 on different boxes? If so, how do their sip.conf files differ? How is your SIP provider sending DTMF to you? SIP INFO, RFC2833 or inband? Running the 1.4 setup, can you create a simple dialplan IVR to test the reception of digits? Or even just a Read, and then echo the results using a NoOp? The point is to see whether the problem is specific to Meetme or general within your 1.4 build of Asterisk. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: problems using the 1.4 version of meetme
The 1.4 and 1.2 are alternately on the same box -- I have to install 1.2 rmmod the zaptel modules delete /usr/lib/asterisk/modules/* and install the zaptel and asterisk. I use the same procedure to go back to the 1.4. I know the ivr works because I have to use a ivr menu and even enter a password to get to the meetme conference and those work fine. The sip provider is using inband as I have requested. Also, I tried calling through the sip provider to my local extension and I hear the tones just fine, so its a mystery to me. on Friday 12/22/2006 Tony Mountifield([EMAIL PROTECTED]) wrote In article [EMAIL PROTECTED], John covici [EMAIL PROTECTED] wrote: Hi. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do the exacct same procedure but use a phone number which calls my box via a zap channel -- using a digium card -- it works perfectly. This problem seems to be independent of asterisk 1.4 and zaptel 1.4 versions, but I did an svn update this morning on both of those. Now this problem does NOT occurr with 1.2 at all, I can call my box using sip and the * is seen by the meetme conference. Are the 1.2 and 1.4 on different boxes? If so, how do their sip.conf files differ? How is your SIP provider sending DTMF to you? SIP INFO, RFC2833 or inband? Running the 1.4 setup, can you create a simple dialplan IVR to test the reception of digits? Or even just a Read, and then echo the results using a NoOp? The point is to see whether the problem is specific to Meetme or general within your 1.4 build of Asterisk. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] Re: problems using the 1.4 version of meetme
In article [EMAIL PROTECTED], John covici [EMAIL PROTECTED] wrote: The 1.4 and 1.2 are alternately on the same box -- I have to install 1.2 rmmod the zaptel modules delete /usr/lib/asterisk/modules/* and install the zaptel and asterisk. I use the same procedure to go back to the 1.4. I know the ivr works because I have to use a ivr menu and even enter a password to get to the meetme conference and those work fine. The sip provider is using inband as I have requested. Also, I tried calling through the sip provider to my local extension and I hear the tones just fine, so its a mystery to me. Do you have a SIP phone you can register directly with the box and try? If so, try setting it to the three different ways of sending DTMF and see whether any of them work. Just trying to whittle down the possibilities to start with You may find it better to use out-of-band DTMF with SIP. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: problems using the 1.4 version of meetme
OK, this brings up a possible lack of understanding by me regarding the dtmf and sip relationship. I see a dtmfmode in the sip.conf and it says the mode for sending dtmf -- rfc2833 info or inband . Now what I don't understand is what controls what asterisk is looking for in terms of dtmf -- does it only look for the one in dtmfmode? this may help solve the problem or at least clarify some thing. Thanks. on Friday 12/22/2006 Tony Mountifield([EMAIL PROTECTED]) wrote In article [EMAIL PROTECTED], John covici [EMAIL PROTECTED] wrote: The 1.4 and 1.2 are alternately on the same box -- I have to install 1.2 rmmod the zaptel modules delete /usr/lib/asterisk/modules/* and install the zaptel and asterisk. I use the same procedure to go back to the 1.4. I know the ivr works because I have to use a ivr menu and even enter a password to get to the meetme conference and those work fine. The sip provider is using inband as I have requested. Also, I tried calling through the sip provider to my local extension and I hear the tones just fine, so its a mystery to me. Do you have a SIP phone you can register directly with the box and try? If so, try setting it to the three different ways of sending DTMF and see whether any of them work. Just trying to whittle down the possibilities to start with You may find it better to use out-of-band DTMF with SIP. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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
