[Spce-user] spce caller name peered with Asterisk/FreeSwitch

[Jon Bonilla]

El Wed, 10 Aug 2011 12:03:19 -0400
Jonathan Scott <jonathan at xpressamerica.net> escribió:

> On 8/10/2011 11:51 AM, Jon Bonilla wrote:
> >
> >> I have pasted the logs here: http://pastebin.com/QduSXbfB
> >> The example call is 2073911000 ->  2073911666, theres only a few subs on
> >> there so the logs aren't too busy.
> >>
> >
> > Please keep the list in CC
> >
> > What's the problem with the call? What do you expect to happen that it's not
> > happening?
> >
> >
> > Jon
> >
> Hi, thanks for the reply I apologize for not CC'ing the list.
> 
> That particular call is placed from our Asterisk box, the caller name 
> should be XpressAmerica, not 207391100 at 71.241.208.21 which is the IP of 
> the Asterisk Box.
> Am I wrong to expect SPCE to retain caller name information? I've never 
> had issues with CNAME in a pure sip environment... our Asterisk box gets 
> it's caller information off of a PRI, and I can understand why that may 
> not be presented to that particular trunk at the moment, but those calls 
> are placed directly off the Asterisk box from IP phones for the purpose 
> of confirming CNAME is being presented.
> 
> 
> 
> 


If I understood well your log, this is the INVITE request the sip:provider
receives:

U 2011/08/10 11:28:07.411716 71.241.208.21:5060 -> 71.241.208.11:5060
INVITE sip:2073911666 at sip.xatel.net SIP/2.0'
Via: SIP/2.0/UDP 71.241.208.21:5060;branch=z9hG4bK26176ad1;rport'
From: "2073911000" <sip:2073911000 at 71.241.208.21>;tag=as0a2c7bf8'
To: <sip:2073911666 at sip.xatel.net>'
Contact: <sip:2073911000 at 71.241.208.21>'
Call-ID: 4339e26b53cbf14c682e3869679f6c46 at 71.241.208.21'
CSeq: 102 INVITE'
User-Agent: FPBX-2.8.1(1.4.23.1)'
Max-Forwards: 70'
Date: Wed, 10 Aug 2011 15:28:07 GMT'
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY'
Supported: replaces'
Content-Type: application/sdp'
Content-Length: 287'
'
v=0'
o=root 2953 2953 IN IP4 71.241.208.21'
s=session'
c=IN IP4 71.241.208.21'
t=0 0'
m=audio 12504 RTP/AVP 0 8 3 101'
a=rtpmap:0 PCMU/8000'
a=rtpmap:8 PCMA/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-16'
a=silenceSupp:off - - - -'
a=ptime:20'
a=sendrecv'



The from header you asterisk box sends is 
"2073911000" <sip:2073911000 at 71.241.208.21>

There's no "XpressAmerica" in that INVITE.
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