[Asterisk-Users] call waiting beep on PSTN and TDM400P FXO line hook flash
I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have two FXS ports on the TDM400P. So my problem is, how do I flash the Verizon PSTN line when I hear the call waiting beep? How can I send a hook flash to the Verizon trunk to activate their 3-way calling feature. I have seen some stuff like hook flash then send *0 to get the bridged Zap trunk to flash but I can't get it to work. I get the reorder signal. I need something my wife and kid can do. Can anybody help? Thanks, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Waiting Beep
All, I was just informed by someone that they can hear the call waiting beep on my phone when I get a second call. Has anyone seen this before, and is there a way to prevent it? I'm using a Cisco 7940 connected to an * box, which is connected to a channel bank that terminates my POTS lines (incase that makes any difference). TIA, Kenny signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call waiting beep
Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the in the dial statement. i.e.) exten = blah,blah,Dial(SIP/GS1SIP/GS2SIP/GS3SIP/ata186aSIP/ata186b,25,t) If one of those lines is being used, then the user gets a really loud call waiting beep, and on the ata186, also an inband callerid noise (perhaps changeable on the ata186). thanks for any info. Sean Rodger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting beep
Sean Rodger wrote: Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the in the dial statement. i.e.) exten = blah,blah,Dial(SIP/GS1SIP/GS2SIP/GS3SIP/ata186aSIP/ata186b,25,t) If one of those lines is being used, then the user gets a really loud call waiting beep, and on the ata186, also an inband callerid noise (perhaps changeable on the ata186). thanks for any info. Sean Rodger The loud call waiting noise from the GS is supposed to be resolved in the next firmware release.. Hopefully it will be out soon.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting beep
Sean Rodger wrote: Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the in the dial statement. i.e.) exten = blah,blah,Dial(SIP/GS1SIP/GS2SIP/GS3SIP/ata186aSIP/ata186b,25,t) If one of those lines is being used, then the user gets a really loud call waiting beep, and on the ata186, also an inband callerid noise (perhaps changeable on the ata186). thanks for any info. Sean Rodger The loud call waiting noise from the GS is supposed to be resolved in the next firmware release.. Hopefully it will be out soon.. Later.. Until GS comes out with their release, you can try the patch I put up on http://bugs.digium.com/bug_view_page.php?bug_id=408 There is a problem with that Michael pointed out to me, re holding and using ChanIsAvail which I'm looking into right now, but it works for normal incoming/outgoing calls via Dial as per your scenario. Check it out and let me know how you go. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
