Thanks for the help,
Without access to our Avaya CM, I've tried to implement the following rules in my config (as per the nice book "Building Telephony Systems with OpenSIPS") (see below) After restart of the Opensips server , and fixing a few things... I can see the following in the logfile: Mar 30 09:11:28 sip /sbin/opensips[15670]: new branch at sip:[email protected] Mar 30 09:18:57 sip /sbin/opensips[15673]: new branch at sip:[email protected] (I'm assuming this means my sip server is trying to forward the request to the PBX, which I know requires a 9 to make outside call) However, Call is not being received by the destination phone # I was 'told" that Avaya CM requires an SES server in front of it to talk to other 3rd Party SIP servers (i.e. OpenSIPS). Any truth to this? All I want for now is to make outbound calls... Any other debug messages I can enable / look for? == config added to my SIPs server== sip:/etc/opensips # diff opensips_backup_3_30_2010.cfg opensips.cfg 87a88,90 > loadmodule "permissions.so" > loadmodule "group.so" > 163a167,172 > # ----- Group ---- > modparam("group","db_url","mysql://XXXX:xx...@localhost/opensips") > > # ----- Permissions ---- > #modparam("permissions","db_mode",1) > modparam("permissions","db_url","mysql://XXXX:x...@localhost/opensips") 273a283 > if(!check_source_address("0")){ 284a295 > } 364a376,414 > #--- Routing to the PSTN section ---# > if (uri=~"^sip:[2-9][0-9]{6}@") { > #Normalize the local number e.164-Miami(1305) > if (db_is_user_in("credentials","local")) { > prefix("91408"); > route(4); > exit; > } > else { > sl_send_reply("403", "No permissions for local calls"); > exit; > }; > }; > > if (uri=~"^sip:1[2-9][0-9]{9}@") { > if (db_is_user_in("credentials","ld")) { > route(4); > exit; > } > else { > sl_send_reply("403", "No permissions for long distance"); > exit; > }; > }; > > if (uri=~"^sip:011[0-9]*@") { > #Normalize for e164 > if (db_is_user_in("credentials","int")) { > strip(3); > route(4); > exit; > } > else { > sl_send_reply("403", "No permissions for internat. calls"); > exit; > }; > }; > > 384a435,439 > route[4] { > #---- PSTN route ----# > rewritehostport("172.16.217.242"); > route(1); > } From: [email protected] [mailto:[email protected]] On Behalf Of Christian Vo Sent: Friday, March 19, 2010 11:29 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] newbie question -- anyone have OpenSIPS working with an AVAYA PBX? At this point, I can make basic SIP calls (i.e. xlite to xlite) =) No experience with dialplan at this point, so I'm curious to know if anyone has OpenSIPS working with Avaya CM? If so, any generic config/ tips that can be shared? Thx ________________________________ This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. ________________________________
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