search for: hasvoicemail

...hanks in advance Regards Bie below is my sip.conf allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm I also had 2 extensions (me at 250 and 998 is my SPA-3102) and my users.conf goes below: [general] fullname=New User userbase=6000 hasvoicemail=yes vmsecret=1234 hassip=yes hasiax=yes hasmanager=no callwaiting=yes threewaycalling=yes callwaitingcallerid=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 host=dynamic localextenlength=0 allow_aliasextns=no allow_an_extns=no hasagent=no hasdirectory=no [...

..." context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XXXXXX] callwaiting = yes cid_number = 951XXXXXX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XXXXXX secret = 000000 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes

...wrong? Some additional information about the configuration: The asterisk version is 1.4.10 ----------------------------------------------------------------------------- In users.conf I defined a user 6002: [6002] fullname = Test Agent email = test.agent at agent.com secret = 1234 zapchan = 1 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = international host=dynamic ----------------------------------------------------------------------------- In agents.conf I added the agent agent => 6002,1234,Test Agent ------------------------...

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...pp_voicemail.so" and "app_hasnewvoicemail.so" loaded (I have "autoload=yes" in modules.conf and have "noload=" a bunch, but I even explicitly set "load=app_voicemail.so" just in case. However, "core show applications like voi" only lists "HasVoicemail" and "HasNewVoicemail", with no sign of "Voicemail". The wiki seems to show that it should all be included, with no sign of deprecation... Any ideas where I can look? TIA, Andrew -- Linux supports the notion of a command line or a shell for the same reason that only chi...

...6_64 running Linux on 2008-10-16 12:37:36 UTC Pbx*CLI> show hints ... 1203 at default : SIP/1203 State:Idle Watchers 0 1202 at default : SIP/1202 State:Idle Watchers 0 ... Users.conf [1202] fullname = 1202 secret = 1202 hasvoicemail = yes mailbox = 1202 at default vmsecret = 1234 hassip = yes hasmanager = no callwaiting = no context = from-nortel call-limit = 4 dynamic = yes qualify = yes host = dynamic [1203] fullname = 1203 secret = 1203 hasvoicemail = yes mailbox = 1203 at default vmsecret = 1234 hassip = yes hasmanager =...

...errors but when i turn on sip debugging it just say unauthorized. below is my users.conf file [6058] allow = all callwaiting = yes cid_number = 6058 context = default email = user@mydomain.com fullname = Bob smith group = hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host=dynamic mailbox = 6058 secret = myPassword threewaycalling = yes vmsecret = 1337 zapchan = registeriax = no registersip = yes Any idea why this doesn't work but registering normal SIP configured accounts does? -------------- next part -------------- An HTML attachment was scrubbed....

...aflags mailbox regexten regserver fromdomain testers.com videosupport no contactpermit contactdeny fullname 660 win8 hasvoicemail subscribemwi dtlsenable yes dtlsverify no dtlscertfile /etc/asterisk/keys/asterisk.pem dtlsprivatekey /etc/asterisk/keys/asterisk.pem dtlssetup actpass sippassw...

...gs 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 = jsmall at visinet.com.au 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 =...

...Blind transfer ;disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record (a.k.a. Touch Monitor) atxfer => * ; A users.conf: [6004] fullname = Analog User 4 secret = 6004 email = cid_number = zapchan = 4 context = numberplan-local hasvoicemail = yes hasdirectory = yes hassip = no hasiax = no hasmanager = no callwaiting = no threewaycalling = no mailbox = 6004 hasagent = no group = 2

...appreciated, but is not the focus of this post: Users.conf ;several handsets setup like this... [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = XXXXX threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 vmsecret = 1234 ;some PSTNS [trunk_2] callerid = asreceived context = DID_trunk_2 group = 2 hasexten = no hasiax = no hassip =...

...box: NULL regexten: NULL fromdomain: testers.com fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sipp...

...mailbox: NULL regexten: NULL fromdomain: testers.com fromuser: 660 qualify: NULL defaultip: NULL outboundproxy: 1.1.1.1 contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sipp...

...appreciated, but is not the focus of this post: Users.conf ;several handsets setup like this... [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = XXXXX threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 vmsecret = 1234 ;some PSTNS [trunk_2] callerid = asreceived context = DID_trunk_2 group = 2 hasexten = no hasiax = no hassip =...

...sip = no host = iax2.fwdnet.net port = 4569 registeriax = yes registersip = no secret = rycort4e trunkname = Custom - fwd trunkstyle = customvoip username = 788694 [6000] callwaiting = yes cid_number = 6000 fullname = proton hasagent = yes hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 6000 secret = proton threewaycalling = yes vmsecret = 1234 registeriax = no registersip = yes canreinvite = yes nat = no dtmfmode = inband disallow = all allow = all context = numberplan-custom-1 *******part of users.conf that was added by asterisk-gui (svn)******* t...

Hi I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. 302 => 302,Website Sales,sip@example.com,,attach=yes|delete=yes However I can't seem to find a way to test is a message is left. I have tried

...usereqphone: NULL textsupport: NULL faxdetect: NULL buggymwi: NULL auth: NULL fullname: NULL trunkname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL autoframing: NULL rtpkeepalive: NULL call-limit: NULL g726nonstandard: NULL ignoresdpversion: NULL allowtransfer: NULL dynamic: NULL path: NULL supportpath: NULL sippasswd...

Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can

...ick up after 2 rings rather than 4? (that is, if you knew how to turn it on - that required to know how to set the time on your VCR to avoid the flashing 12:00:00) Hahaha. Jokes aside. I have come up with a way to do this but it's a kludge: 1 - Read in the variables (normal_ring_timeout) 2 - HasVoiceMail_Timeout 3 - run HasVoicemail 4 - If not have voicemail set ring_timeout = normal_ring_timeout 5 - If Have voicemail - it's jumps to priority + 101 6 - Set ring_timeout = hasvoicemail_timeout 7 - Ring extensions for ring_timeout 8 - blah blah blah... I was wondering if someone had come up with...

...onf is emply extension.conf exten = 66666,1,Answer() exten = 66666,2,Queue(${EXTEN}) and users.conf with users.conf [60011] callwaiting = yes cid_number = 60011 context = numberplan-custom-2 fullname = Roberto Bolivar Casa hasagent = yes hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes deletevoicemail = no host = dynamic mailbox = 60011 secret = 123654 threewaycalling = yes vmsecret = 1234 registeriax = no registersip = yes canreinvite = no nat = yes dtmfmode = rfc2833 disallow = all allow = all the client Xlite call to 66666 and the enter to musicold and not pass to the...