[asterisk-users] Sending SIP 183 Session Progressing

[Michael J. Tubby B.Sc (Hons) G8TIC]

Dinesh,

I suspect your problem is with the softphone implementation...

I have an Asterisk PBX setup with ISDN (chan_capi) and use Cisco 7960 phones 
with Cisci SIP 7.5 firmware and get to watch the various SIP messages in/out 
on the phone.

Depending on the phone numbers I dial (and the signalling back from the ISDN 
exchange) I get 100 -> 183 -> 180 or 100 -> 180

In both cases the Cisco plays our ringing on receipt of the 180.

Occasionally calls which go from 100 -> 180 without going via the 183 result 
in the Cisco ringing and combined rining genrated by the telephone exchange 
which is weird but ok.

I have also encountered (rarely) ISDN number which, when dialled from 100 -> 
183 -> Connected without a ringing phase - these call result in silence at 
the Cisco phone followed by connected audio (from the far end) - which is to 
be expected.


Mike




----- Original Message ----- 
From: "Dinesh Nair" <dinesh at alphaque.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Tuesday, August 15, 2006 7:18 AM
Subject: [asterisk-users] Sending SIP 183 Session Progressing


>
> i'm not sure if this is a -users or a -dev question, but am sending it 
> here anyways. discussion could move to -dev if chan_sip.c code needs to be 
> amended/explained.
>
> first up, all this on asterisk 1.2.10 on freebsd 6.1.
>
> here's the beef:
>
> from a particular sip softphone we're playing with, we notice that calls 
> to another SIP phone (same LAN) result in the /lack/ of a ringing tone on 
> the softphone. however, calls from the same softphone to a PSTN/Mobile 
> number (through a TE405P) result in proper behaviour on the softphone with 
> a ringing tone.
>
> an ethereal trace of both types of calls results in only one difference. 
> for calls to the PSTN/Mobile thru libpri/chan_zap, asterisk returns a SIP 
> 183 Session Progress[1] packet in between the 100 Trying and 180 Ringing, 
> while for calls from the softphone to another SIP phone it's 100 Trying 
> followed immediately by 180 Ringing.
>
> so my question is, is the softphone behaving correctly in not playing a 
> ringing tone to the user without the 183 packet inspite of the 180 Ringing 
> packet being received ? alternatively, since we aren't able to change the 
> softphone, will i break anything big if i force asterisk to send the 183 
> packet immediately after sending the 100 Trying packet in sip_indicate() ?
>
> alternatively, in reading the RFCs, i came across RFC3398 which speficies 
> mappings between ISDN Cause Codes and SIP responses. has this mapping been 
> implemented in asterisk at the moment, either in 1.2 or the upcoming 1.4 ?
>
> [1] the 183 Session Progress packet is triggered by the receipt of a PRI 
> PROGRESS indicator from libpri, which gets translated to a 
> AST_CONTROL_PROGRESS and thence a 183 Session Progress to SIP.
>
> -- 
> Regards,                           /\_/\   "All dogs go to heaven."
> dinesh at alphaque.com                (0 0) 
> http://www.openmalaysiablog.com/
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