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(${HANGUPCAUSE})
[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 mcolo...@enter.it
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
http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature
-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,
Someone can tell me if asterisk support the SIP History-Info?
If it supports, how can enable it?
I searched on Google
-users-boun...@lists.digium.commailto:asterisk-users-boun...@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 10:45 AM
To: Asterisk Users Mailing List - Non
: 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 within chan_sip.
Cheers,
--
Joshua Colp
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, H,
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 same
