You need to log your agents in - or set your queue members to be SIP accounts. (which is probably the best solution)
PaulH On Wed, 2007-09-05 at 16:53 +1000, Joshua Small wrote: > Hi, > > I’ve just built my first asterisk server. Current information: > > > > OS Version: > > Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 > 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux > > > > Asterisk Build: > > Asterisk 1.4.11 > Asterisk GUI-version Revision: 1479 $ > > > > Server Date & TimeZone: > > Thu Sep 6 02:37:11 EST 2007 > > > > I’ve used the Asterisk GUI for setup with two IP handsets, one VOIP > account with a telco and one PSTN. The server correctly allows: > > - Handsets to call each other > > - Calls outbound through both PSTN or VOIP > > > > I’m having an issue with incoming calls however. If I configure > “incoming calls” coming over my PSTN to a single user, it works > correctly (that handset rings, can pickup etc). However if I define a > call queue which consists of both these handsets, neither ever rings. > > > > Looking at the console, I see this: > > -- Started music on hold, class 'default', on Zap/1-1 > > [Sep 6 02:22:51] WARNING[5955]: channel.c:2129 ast_waitfordigit_full: > Unexpected control subclass '2' > > [Sep 6 02:22:54] WARNING[5955]: channel.c:2129 ast_waitfordigit_full: > Unexpected control subclass '2' > > > > The error repeats until the caller hangs up. > > > > I’ve posted all the config that I felt was relevant here, let me know > if you need more. This was all written by Asterisk-GUI. I realise > there’s a lot more configuration but given that things work fine when > I set the receive to a single agent, I assumed it was a queue issue. > > > > Users.conf > > [6001] > > callwaiting = yes > > context = numberplan-custom-1 > > email = [EMAIL PROTECTED] > > fullname = Joshua Small > > hasagent = yes > > hasdirectory = yes > > hasiax = no > > hasmanager = no > > hassip = yes > > hasvoicemail = no > > host = dynamic > > mailbox = 6001 > > secret = SECRET > > threewaycalling = yes > > registeriax = no > > registersip = yes > > canreinvite = no > > nat = no > > dtmfmode = rfc2833 > > > > > > Queues.conf > > [6003] > > fullname = All of us > > strategy = ringall > > timeout = > > wrapuptime = > > autofill = yes > > autopause = no > > maxlen = > > joinempty = no > > leavewhenempty = no > > reportholdtime = no > > musicclass = > > member = Agent/6001 > > member = Agent/6002 > > > > extensions.conf - broken > > [DID_trunk_2] > > include = default > > exten = _X.,1,Goto(default|6003|1) > > exten = s,1,Goto(default|6003|1) > > > > extensions.conf – works but only sends to a single handset > > [DID_trunk_2] > > include = default > > exten = _X.,1,Goto(default|6001|1) > > exten = s,1,Goto(default|6001|1) > > > > Any assistance appreciated. > > > > Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 > 959 | www.visinet.com.au > > This e-mail is intended for use by the named recipients only and > contains confidential information. Opinions and other information in > this message that pertain to the sender's employer and its products > and services represent the opinion of the sender and not > necessarily those of the employer. > > > > > _______________________________________________ > --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 _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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