On Thursday 17 January 2008 05:54:57 Patrick Baker wrote:

Sipwise.com might help

Sincerely I don't think SipWise is a good start point.

Hello Patrick,

you may go the following way: Asterisk: Setup Asterisk for realtime integration as described in the guide: Make sure when Asterisk starts (-vvvr) you get the following messages:

asterisk -vvvr == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip == Binding voicemail to mysql/asterisk/voicemail

If you get errors like: find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available

then you might: - have no mysql-client installed on your asterisk machine - you do nat have sufficient access rights on your openser-mysql - your openser-mysql may list on loacalhost instead of its real IP - you might not have compiled asterisk and asterisk-addons with the –with-mysqlclient=/usr option - I remember that some time ago on one machine I needed also to install mysql-devel and mysql-server as asterisk-addons complained a missing module in /usr/bin while compiling

Now create a route for Asterisk in OpenSER: route[5] {

# ----------------------------------------------------------------- # ASTERISK Handler # ----------------------------------------------------------------- xlog("L_INFO","Step 500 Entering ==== Route 5 (Asterisk) ======"); rewritehostport("www.xxx.yyy.zzz:5060"); # INSERT YOUR Asterisk GATEWAY IP ADDRESS and port here route(1); # now use standort route with t_relay }

Now in your main routing logic branch to the asterisk route if (uri=~"^sip:0[1-9].*@.*") { # Dial national PSTN number here: in Germany append_rpid_hf(); # for submitting external number from customers from subscriber table/rpid route(5); # Asterisk route exit; }; Restart OpenSER.

Now goto Asterisk and register you Openser in sip.conf e.g. [openser] host=<your openser host> dtmfmode=rfc2833 type=peer context=default insecure=very nat=never allow=all

and create lines like the following in externsions.conf in your inbound route exten => _0.,1,Set(MYMSN=${CALLERID(number)}) exten => _0.,n,NoOP(${MYMSN} ${CALLERID(name)} ${EXTEN}) exten => _0.,n,Playback(beep) exten => _0.,n,Hangup _0. means to catch all numbers which start with 0. Restart or reload Asterisk. When a connection is made you shopuld hear a beep.

I hope this helps.

Best regards Peter

Patrick Baker schrieb:

Sipwise.com might help

----- Original Message ----- From: users-bounces@lists.openser.org users-bounces@lists.openser.org To: users@lists.openser.org users@lists.openser.org Sent: Wed Jan 16 23:48:27 2008 Subject: [OpenSER-Users] openSER Call Route Logic

Hi,

I recently installed openser-1.3.0 and able to connect my cisco sip phones and everything is working. Now i need to integrate this with asterisk box for call forwarding to other locations & voicemails. i read the getting started guide and other information, but seems i need more help from the group. can any one give me a helping hand. i am sure i will do my home work before bug you again. I need a starting point to over come the complexity of the routing plan and logic.

thanks -asterisk user

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