Joseph, I have been completely unsucessful in doing the same thing as you - do you use a Mitel LX or MXe controller? We have a few probs with the LX internally too...
Cheers Alex On Wed, 2007-07-18 at 06:42 +0100, Joesph O wrote: > Good morning, it now works, failure was due to > a misconfigured/misunderstood Class of Restriction Group Assignment > for the SIP Trunk Routes on the 3300ICP. > > Now Asterisk can call the world through the Mitel and incoming calls > (DID, operator transfers etc) to Asterisk via the 3300ICP, all work. > > Interesting side note - both phone systems have same range of > extensions e.g 100 - 299 (just an example)and we created routes on > both to point to the other for the range, of course, an > extension should only exist on one at any time. therefore, if an > extension does not exist locally, it is routed to the other and vice > versa, this way, we keep same range of extensions and this has helped > with migrating users who do not want to trade their loved Extension & > DID number for anything. will continue to test and share results. > > Joesph O. > > > On 7/9/07, Joesph O <[EMAIL PROTECTED]> wrote: > Good day everyone, > > I have Asterisk and Mitel 3300 ICP communicating via SIP. > Calls to and from extensions on both sides are completing > successfully (details on config coming soon). Problem is that > calls from Asterisk to PSTN via E1 on Mitel 3300 ICP are > rejected. What do I need to do to get Asterisk to route PSTN > calls through it successfully? > > Here is an extract of the log on Asterisk whenever I try to > call PSTN through 3300ICP, in this case, Extension 2540 on > Asterisk called 2345678, 9 is a leading digit - > > Jul 7 16:48:07 DEBUG[6860] chan_sip.c: Outgoing Call for > 92345678 > Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Called > Mitel3300ICP/92345678 > Jul 7 16:48:07 DEBUG[6860] channel.c: Prodding channel > 'SIP/2540-b7904a98' > Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Started music on > hold, class '24', on SIP/2540-b7904a98 > Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 160 > sample intervals > Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping > retransmission (but retaining packet) on > '[EMAIL PROTECTED]' Request 102: > Found > Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Acked pending invite > 102 > Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Stopping retransmission > on '[EMAIL PROTECTED]' of Request > 102: Match Found > Jul 7 16:48:07 DEBUG[6860] channel.c: Generator got voice, > switching to phase locked mode > Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 0 > sample intervals > Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping > retransmission (but retaining packet) on > '[EMAIL PROTECTED]' Request 103: > Found > Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Acked pending invite > 103 > Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Stopping retransmission > on '[EMAIL PROTECTED]' of Request > 103: Match Found > Jul 7 16:48:08 WARNING[6270] chan_sip.c: Forbidden - wrong > password on authentication for INVITE to '"Tester" < > sip:[EMAIL PROTECTED] >;tag=as07fef065' > Jul 7 16:48:08 VERBOSE[6860] logger.c: -- > SIP/Mitel3300ICP-0832de50 is circuit-busy > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 1200 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users