Happy to say that we’ve successfully set up Asterisk 11 or higher with TM’s Multi-Line SIP which basically uses IMS signalling on Huawei devices used by Telekom Malaysia. We had to modify chan_sip.c and parser files to support TEL: URI for INVITE messages. Currently, we have enabled it to support incoming INVITES only.
TM doesn’t require to send TEL: uri for outgoing calls and the usual SIP: uri is perfectly fine. There are several steps involved and i will blog about it later (when i have the time). Generally its
1) Getting the hack from here: http://forums.asterisk.org/viewtopic.php?f=1&t=76432
2) Adding one or two more TEL support in the parser file
3) Configure trunks and registration
4) Setup an incoming dialplan to chomp down parts of the SIP header to be used as CallerID and DID values respectively.
5) Enable ringing into all inbound routes We successfully tested incoming, outgoing, transfers using standard codecs.
The audio quality is nearly as good as PRI tho sometimes, takes a bit longer to handshake the INVITE messages but it’s hardly noticeable. We might be able to send messages too over regular IP or SMS, i think that’s why the IMS is chosen in the first place, to enable multimedia over VoIP protocols.
If you need help, write to us [email protected] and if you use Asterisk in a non-commercial environment, i will set it up for free.
Astiostech and ORENCloud offers simple solutions to switch to MLS or go cloud with MLS, read more here.
For more information on TM’s MLS: https://www.tm.com.my/Office/Business/SME/Solutions/Pages/Multi-Line-SIP.aspx