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...t;CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind regards, Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/f48b9334/attachment.htm

Hi, How can I strip + from the front of the caller ID? I have tried this: exten => s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1}) But it is not working. Szasz Szabolcs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100114/a3f18781/attachment.htm

Hi! I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed by the clients ? Thanks a lot, Szasz Szabolcs

.... For every call I receive on this trunk, I get the message "That is not a valid conference number". I'm using Asterisk version 1.4.22, I had install the dahdi-linux and dahdi-tools and the conference is working between the phones registered to Asterisk PBX. What's wrong? Thanks. Szasz Szabolcs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090209/58c4fdce/attachment.htm

Hi, How can I disable comfort noise on Asterisk? Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100129/3b27a653/attachment.htm

...ws : http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3 . My voicemail system is working but, I can't get to work the message waiting indicator. It doesn't seems to send the Asterisk any NOTIFY message to the Openser box. How can I debug it? Thank you Szasz Szabolcs

Hi all, I am using asterisk 1.4.25.1. when I am sending dtmfs the first digit is not detected. Do you know a workaround for this? Besst regards, Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110520/b3e29d9f/attachment.htm>

...5060)-event) (5060)g729 telephone-event) (5060)g729 telephone-event)1(5060)1U telephone-event) (5060) <------------------n (5060)ot Existl(5060)-event) (5060)g729 telephone-event) (5060)g729 telephone-event)1(5060)1U telephone-event) I am using version 1.4.25.1 Thank you in advance Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100211/8c710ae0/attachment.htm

...105.199:5051;user=phone;transport=tls> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003 Content-Length: 0 Thanks in advance Szasz Szabolcs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091002/4a450177/attachment.htm

Hy all! I want to play in my radio(Icecast2 + ices0.4)an existing live stream. How can i do? for example: The live stream is http://slageronline.neteyes.hu:9000/ My server is: http://127.0.0.1/live , and i want to play the "http://slageronline.neteyes.hu:9000/" on the "http://127.0.0.1/live" sorry for the bad english... -- This message was scanned for spam and viruses by