[asterisk-users] R: Asterisk and Call Hold

Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value "a=sendrecv" is present, according to the rfc3264 the sdp value a must be mark with "sendonly". I've already tried with Asterisk 1.8 and Asterisk 11, but there is the sa

[asterisk-users] Call Hold problem

Hello everybody, i have a problem with asterisk 1.8 and Call Hold My problem is that Asterisk don't send re-invite when i pick up the call from hold. I already insert canreinvite=no in all my sip channels, set dtmfmode=info in sip.conf and my Dial() command don't insert option like t, T", "h", "

[asterisk-users] R: R: R: Asterisk and History-Info

: Re: [asterisk-users] R: R: Asterisk and History-Info Marco Colombo wrote: > Hi, Hola, > On my invite trace I don't have history-info. > > Could you explain me how do I put "history-info" on SIP INVITE? You can't. That specific RFC (4244) is not implemented with

[asterisk-users] R: R: Asterisk and History-Info

t; [mailto:asterisk-users-boun...@lists.digium.com]<mailto:[mailto:asterisk-users-boun...@lists.digium.com]> On Behalf Of Marco Colombo Sent: Wednesday, September 26, 2012 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: Asterisk and History-Inf

[asterisk-users] R: Asterisk and History-Info

oun...@lists.digium.com> [mailto:asterisk-users-boun...@lists.digium.com]<mailto:[mailto:asterisk-users-boun...@lists.digium.com]> On Behalf Of Marco Colombo Sent: Wednesday, September 26, 2012 10:33 AM To: Asterisk-Users Subject: [asterisk-users] Asterisk and History-Info Hi All,

[asterisk-users] Asterisk and History-Info

Hi All, Someone can tell me if asterisk support the SIP History-Info? If it supports, how can enable it? I searched on Google, but I could not find anything... Thanks for all Best Regards MC

[asterisk-users] R: SIP CANCEL, Reason

[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matthew Jordan Inviato: giovedì 20 settembre 2012 13:42 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] SIP CANCEL, Reason - Original Message - > From: "Marco Colombo" &

[asterisk-users] SIP CANCEL, Reason

Hi All! i have a problem with asterisk 1.8.11. I must have in the SIP cancel message, the line "Reason" Example : Reason : SIP;cause=16;text="Normal Call Clearing" I have already enable "use_q850_reason=yes", but this not work. In my dialplan I have already add : exten => _X.,n,Hangup(${HANGUPCAU