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

Jon Bonilla jbonilla at sipwise.com
Wed Aug 10 04:12:34 EDT 2011

El Tue, 09 Aug 2011 18:02:16 -0400
Jonathan Scott <jonathan at xpressamerica.net> escribió:

> Hello list,
> I have SPCE peered with Asterisk & FreeSwitch for redudancy, among other 
> things..
> We're only passing PSTN traffic to SPCE, and do not allow calls from 
> outside sip domains.
> SPCE is not pulling the Caller Name information presented by Asterisk, 
> or FreeSwitch..
> Instead prefering CID#@domain.net for the CNAM.
> Caller ID numbering is fine, although I would rather even have the 
> system present simply CID/CID without the domain if it can't present 
> CNAM/CID for now..
> CNAM is a big thing for our subscribers, and I'm extremely anxious to 
> get spce into production. any thoughts?

Could you send a trace (using "ngrep-sip" in the spce) and extract kamailio logs
(/var/log/ngcp/kamailio-proxy.log) for an example call?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20110810/549892bb/attachment-0001.asc>

More information about the Spce-user mailing list