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

[Marco Colombo]

Hi,

On my invite trace I don't have history-info.

Could you explain me how do I put "history-info" on SIP INVITE?

<------------>
    -- Executing [+39xxxxxxxxxxxx at trunk-squire-incoming:1] Dial("SIP/trunk-squire-outcoming-00000045", "SIP/xxxxxxxxxxxx") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11186
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.20.1.2:5060:
INVITE sip:xxxxxxxxxxxx at 10.20.1.2;uniq=73A845E0147AC676B88F6EC07EFF8 SIP/2.0
Via: SIP/2.0/UDP yyyyyyyyyyyyyyy:5060;branch=z9hG4bK56839522;rport
Max-Forwards: 70
From: "+39zzzzzzzzzzz" <sip:+39zzzzzzzzzzz at yyyyyyyyyyyyyyy>;tag=as3e3ef2cf
To: <sip:xxxxxxxxxxxx at 10.20.1.2;uniq=73A845E0147AC676B88F6EC07EFF8>
Contact: <sip:+39zzzzzzzzzzz at yyyyyyyyyyyyyyy:5060>
Call-ID: 4320ac5e6895a1c40e809ee973c7bed6 at yyyyyyyyyyyyyyy:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.9.0-rc1
Date: Wed, 26 Sep 2012 18:37:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "+39zzzzzzzzzzz" <sip:+39zzzzzzzzzzz at yyyyyyyyyyyyyyy>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 239

Thanks a lot!
Marco


Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Danny Nicholas
Inviato: mercoledì 26 settembre 2012 17:48
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] R: Asterisk and History-Info

Versions 1.8 and 11 (probably 10 as well) let you query SIP information. 1.2 and 1.4 (1.6 also I think) do not.  If you are in a small environment, you can turn on SIP debug and put that in a separate log (would eat up the disk in a few days in most real environments).

From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com]<mailto:[mailto:asterisk-users-bounces at 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-Info

Hi,
Thanks for reply
What do you mean with "Using flat or Realtime log files"?
I need this line in the SIP Invite :

History-Info: <sip:+3906330XXXXX at enter.it;user=phone;cause=302;privacy=history>;index=1
History-Info: <sip:+3906330XXXXX at enterSIP/2.0 100 Trying

how can I provide the data that you asked before?

Thanks
Best Regards

Da: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com]<mailto:[mailto:asterisk-users-bounces at lists.digium.com]> Per conto di Danny Nicholas
Inviato: mercoledì 26 settembre 2012 17:34
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] Asterisk and History-Info

That may depend on the flavor of Asterisk you are using and whether you are using flat or realtime log files.

From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com]<mailto:[mailto:asterisk-users-bounces at 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, but I could not find anything...

Thanks for all
Best Regards

MC

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