Ok, I have made some progress debugging this. I dont believe it has anything to do with asterisk or my phone. Rather I think it is an issues with STUN and/or my Linksys router at home.
The phones I am testing all sit behind a NAT'd firewall, your basic Linksys router for the Home DSL user. The phones all of STUN setup, and the STUN server IP is the IP of the asterisk server - which is purely public. I was able to duplicate the problem with not being able to hear the voicemail greeting by doing the following: Turn off all the phones, and power cycle my Linksys, then turn on 1 phone. That one phone will then work, and you can hear voicemail greeting. The I turn on the second phone. Then voicemail greeting breaks, and you cant hear it when you dial into voicemail. If I unplug the first phone, and power cycle the Linksys again, the second phone will begin to work. So the question is, does this behavior make sense? I assumed with an STUN server I could have multiple phones behind my Linksys firewall, now it appears I can only have one. Is it a Linksys bug, or a general known issue? Do I need to run multiple STUN servers? Thanks John On Jan 31, 2008, at 1:00 PM, Shane D wrote: > Very odd. Could you try taking the mailbox line out of sip.conf and > see what happens? > > On 1/31/08, John Von Essen <[EMAIL PROTECTED]> wrote: >> Here are my configs: >> >> >> sip.conf: >> >> [general] >> context=default >> bindport=5060 >> bindaddr=0.0.0.0 >> disallow=all >> allow=ulaw >> >> [6000] >> type=friend >> secret=letmein >> host=dynamic >> dtmfmode=rfc2833 >> mailbox=6000 >> context=default >> >> extensions.conf: >> >> [default] >> exten => 1000,1,Ringing >> exten => 1000,2,Wait(2) >> exten => 1000,3,VoicemailMain >> >> Calling from phone to phone is fine, and inbound and outbound calling >> is fine. But when I call voicemail, I dont hear anything. >> >> When I view console in CLI I see this when attempting to dial the >> voicemail extension: >> >> -- Executing [EMAIL PROTECTED]:1] Ringing("SIP/6001-081d65c8", "") in >> new stack >> -- Executing [EMAIL PROTECTED]:2] Wait("SIP/6001-081d65c8", "2") in >> new >> stack >> -- Executing [EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081d65c8", >> "[EMAIL PROTECTED]") in new stack >> -- <SIP/6001-081d65c8> Playing 'vm-login' (language 'en') >> [Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate: >> Couldn't read username >> Really destroying SIP dialog '[EMAIL PROTECTED]' Method: >> BYE >> >> So it plays the greetings, and is working, I just cant hear it. >> >> -john >> >> >> >> >> >> On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote: >> >>> On Jan 31, 2008 12:30 AM, John Von Essen <[EMAIL PROTECTED]> wrote: >>>> >>>> Any ideas what could be going on? I tried tweaking the extension >>>> 1000 >>>> so it looks like: >>> >>> Maybe the SIP config is wrong? >>> >>>> >>>> Where 6000 is my mailbox. But still nothing, when I dial 1000, it >>>> just >>>> goes silent. >>> >>> Can you places other calls from that new phone? >>> >>>> Please help. This is driving me nuts. I even tried re-installing >>>> asterisk from scratch - no change. >>> >>> What version? >>> >>> _______________________________________________ >>> -- 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 >>> >> >> >> _______________________________________________ >> -- 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 >> > > > -- > -Shane > Blog: http://blind-geek.com/blog/ > CoOwner: http://sjtechzone.com > AIM: inhaddict > Skype: chatter8712 > > _______________________________________________ > -- 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 > _______________________________________________ -- 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