David,
Thanks. I tried the tip but it didn't help. Below is my
extensions.conf file. Do you see anything off? I think my
voicemail problem is something different, since I am successfully
leaving voicemails. I think my real problem is that I'm not getting
any audio back, since my conference rooms and playback functions are
not working either. So, bottom line, I really only have the first
two problems that I mentioned: a) can not receive incoming calls to
asterisk and b) can not get audio from asterisk.
If you or anyone else have any ideas I would appreciate it. I'm
searching around online but am not finding much luck there.
~Ken
;
; Dial Plan Configuration
;
[globals]
[general]
static=yes
writeprotect=no
clearglobalvars=no
[default]
[google-in]
exten => s,1,NoOp()
exten => s,n,Set(crazygooglecid=${CALLERID(name)})
exten => s,n,Set(stripcrazysuffix=${CUT(crazygooglecid,@,1)})
exten => s,n,Set(CALLERID(all)=${stripcrazysuffix})
exten => s,n,Answer()
exten => s,n,Ringing()
exten => s,n,Wait(2)
exten => s,n,SendDTMF(1)
exten => s,n,Dial(SIP/1000,20)
[macro-voicemail]
exten => s,1,Dial(SIP/${ARG1},15,tT)
exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
exten => s,n(unavail),Voicemail(${MACRO_EXTEN}@default,u)
exten => s,n,Hangup()
exten => s,n(busy),Voicemail(${MACRO_EXTEN}@default,b)
exten => s,n,Hangup()
[macro-conference]
exten => s,1,MeetMeCount(${ARG1},CONFCOUNT)
exten => s,n,GotoIf($[${CONFCOUNT} <= 10]?meetme:conf_full,1)
exten => s,n(meetme),MeetMe(${ARG1},i)
exten => s,n,Playback(conf-thereare)
exten => s,n,MeetMeCount(${ARG1})
exten => s,n,Playback(conf-peopleinconf)
exten => conf_full,1,Playback(conf-full)
[talk-gmail-outbound]
exten => _[a-z][email protected],1,Dial(Gtalk/gmail/${EXTEN}@gmail.com)
[talk-numeric-outbound]
exten => _91NXXNXXXXXX,1,Dial(Gtalk/gmail/+${EXTEN:1}@voice.google.com)
[internal]
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,SendDTMF(1)
exten => s,n,Background(enter-ext-of-person)
exten => s,n,WaitExten()
;lines of individuals
exten => 1000,1,Macro(voicemail,1000)
exten => 1001,1,Macro(voicemail,1001)
exten => 1002,1,Macro(voicemail,1002)
exten => 1003,1,Macro(voicemail,1003)
exten => 1004,1,Macro(voicemail,1004)
exten => 1005,1,Macro(voicemail,1005)
;conference rooms
exten => 600,1,Macro(conference,600)
exten => 601,1,Macro(conference,601)
exten => 602,1,Macro(conference,602)
exten => 603,1,Macro(conference,603)
exten => 604,1,Macro(conference,604)
exten => 605,1,Macro(conference,605)
;voicemail line
exten => 1835,1,VoiceMailMain()
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
[phones]
include => talk-gmail-outbound
include => talk-numeric-outbound
include => internal
include => google-in
On Fri, Dec 7, 2012 at 12:50 PM, David Kerr <[email protected]> wrote:
> Note that on inbound calls from GTalk you need to send a DTMF tone '1' to
> tell google to connect the call. My dialplan is fairly complex but see
> snips below for how I handle this.
>
> I don't know if this will help with your audio problem though.
>
> David
>
>
> DID_gtalk]
> exten = s,1,Set(crazygooglecid=${CUT(CALLERID(name),@,1)})
> exten = s,n,Set(CALLERID(all)=${crazygooglecid})
> exten = s,n,Set(CALLERID(num)=${crazygooglecid})
> ; Calls from GTalk computer clients will have a username as caller ID, not a
> number.
> ; This will cause privacy manager check to fail (which only accepts numeric
> caller ID numbers)
> ; Therefore replace caller ID number with '1234' if it is non-numeric.
> ; This increases risk that anonymous calls through Gtalk will get through
> exten = s,n,ExecIf($[${REGEX("^[+]?[0-9]{4\,}$" ${CALLERID(num)})} !=
> 1]?Set(CALLERID(num)=1234))
> exten = s,n,Set(__GTALKOPTIONS=D(:1))
> exten = s,n,Goto(voicemenu-david,s,1)
> exten = h,1,Hangup()
>
> [gtalk-answer]
> exten = s,1,ExecIf($["${CDR(answer)}"!=""]?Return()) ; Line already answered
> exten = s,n,Answer()
> exten = s,n,Ringing()
> exten = s,n,Wait(2)
> exten = s,n,ExecIf($["${CUT(CHANNEL,/,1)}"="Gtalk"]?SendDTMF(1))
> exten = s,n,Set(__GTALKOPTIONS=)
> exten = s,n,Return()
>
> [voicemenu-david]
> exten => s,1,NoOp(David)
> same => n,GoSub(gtalk-answer,s,1)
> same => n,Set(TIMEOUT(response)=3)
> same => n,Background(record/WelcomeToDavid)
> same => n,WaitExten(1)
> exten => 0,1,VoiceMailMain(101)
> exten => t,1,Voicemail(101,u)
> exten => i,1,Playback(pbx-invalid)
> same => n,Goto(s,4)
> exten => fax,1,Gosub(fax-rx,s,1)
> same => n,Hangup()
> exten => h,1,Hangup()
>
>
> [example]
> ; If you do a Dial() inside your dialplan having not answered (via
> gtalk-answer) the line, then on pickup you must
> ; send a DTMF '1' which is accomplished by the GTALKOPTIONS variable.
> exten => s,1,Dial(SIP/101${softext},${myringtime},${GTALKOPTIONS}it)
>
>
> On Fri, Dec 7, 2012 at 12:35 PM, Ken Fischer <[email protected]>
> wrote:
>>
>> Also..my voicemail stopped working. When I call a line it rings and then
>> when I hit unavail status it seems to transfer me but I hear nothing on the
>> line. When I go to the Astlinux voicemail tab though it shows that I did in
>> fact leave a voicemail.
>>
>> Not sure if this problem is related or not. It started right after I
>> upgraded to 1.8.
>>
>> On Dec 7, 2012 11:53 AM, "Ken Fischer" <[email protected]>
>> wrote:
>>>
>>> Everyone,
>>>
>>> The changes are made and now I am running version 1.8. I now see that
>>> I am connected in the Jabber status and I can use the phone to make
>>> outgoing calls. However, I have the following problems:
>>>
>>> 1. Incoming calls do not work.
>>> 2. During outgoing calls, the voice does not transmit from the
>>> outside line back to the Softphone.
>>>
>>> I suspect these two are related, like I'm only getting one way
>>> communications. Any thoughts?
>>>
>>> Thanks!
>>> ~Ken
>>>
>>> On Wed, Dec 5, 2012 at 12:44 PM, Lonnie Abelbeck
>>> <[email protected]> wrote:
>>> > Ken,
>>> >
>>> > I assume this is on a Compact Flash card or such, be sure to "zero out"
>>> > the drive before re-installing the new image, otherwise the previous ext2
>>> > partition(s) will reincarnate themselves.
>>> >
>>> > Lonnie
>>> >
>>> >
>>> > On Dec 5, 2012, at 11:28 AM, Ken Fischer wrote:
>>> >
>>> >> Thanks....I will get to work on that. The Astlinux box was setup by my
>>> >> professor and is in his office so I will need to coordinate with him. It
>>> >> might be a few days until I can test again and let you know the results.
>>> >> Thank you.
>>> >>
>>> >> On Dec 5, 2012 10:52 AM, "Lonnie Abelbeck" <[email protected]>
>>> >> wrote:
>>> >> Ken,
>>> >>
>>> >> As Michael noted (below) you are running an old version of Asterisk
>>> >> (1.4.44) which I seem to recall has issues with GTalk.
>>> >>
>>> >> I'd suggest using AstLinux 1.0.5 (as you are) but with Asterisk 1.8.x.
>>> >>
>>> >> Using this Link:
>>> >> http://www.astlinux.org/release/105-asterisk-18160
>>> >>
>>> >> Download the "astlinux-1.0.5-asterisk-1.8.16.0.img.gz" for your
>>> >> hardware and install and setup as you have done previously and reapply
>>> >> your
>>> >> edits to the new default configuration.
>>> >>
>>> >> Granted, you have have other issues as well, but you want to start
>>> >> with Asterisk 1.8.x .
>>> >>
>>> >> There is a method to upgrade to Asterisk 1.8.x from your existing
>>> >> setup, but you will still have the Asterisk 1.4 default configs, I think
>>> >> the
>>> >> above reinstall is best for you.
>>> >>
>>> >> Lonnie
>>> >>
>>> >>
>>> >> On Dec 5, 2012, at 6:40 AM, Ken Fischer wrote:
>>> >>
>>> >> > Thank you for your response... much appreciated. I added them to the
>>> >> > modules configuration so I think they are loaded. Is there some other
>>> >> > way to
>>> >> > check? I am using Astlinux version 1.8.
>>> >> >
>>> >> > On Dec 5, 2012 6:42 AM, "Michael Keuter" <[email protected]>
>>> >> > wrote:
>>> >> >
>>> >> > Am 05.12.2012 um 04:06 schrieb Ken Fischer:
>>> >> >
>>> >> > > I'm hoping someone here might be able to help me.
>>> >> > >
>>> >> > > I'm working on configuring astlinux for a school project and I'm
>>> >> > > having some trouble with GTalk integration. My configuration is
>>> >> > > below. I signed up for the astlinux mailing list just so you know
>>> >> > > and
>>> >> > > am waiting to hear back. I can't make outgoing or receive
>>> >> > > incoming
>>> >> > > calls to the phone number associated with
>>> >> > > [email protected] is 484-469-0192. My server
>>> >> > > is
>>> >> > > found at https://mickey.ece.villanova.edu which you can go to and
>>> >> > > see
>>> >> > > my status screen.
>>> >> > >
>>> >> > > I'm noticing that my jabber status says "No such command 'jabber
>>> >> > > show
>>> >> > > connections' (type 'help jabber show connections' for other
>>> >> > > possible
>>> >> > > commands)" This kind of indicates to me that something is wrong
>>> >> > > with jabber and as a result I'm not able to make a connection to
>>> >> > > the
>>> >> > > GTalk services and that is why my incoming/outgoing calls are not
>>> >> > > working.
>>> >> > >
>>> >> > > Is there some way to test to see if I am connecting to GTalk? Do
>>> >> > > you
>>> >> > > see anything wrong in my configuration files related to GTalk?
>>> >> > > I'm at
>>> >> > > a loss right now to explain what is going on.
>>> >> > >
>>> >> > > THANK YOU!
>>> >> >
>>> >> > First: I guess you would get more help in the asterisk-users list.
>>> >> > Anyway: are you sure the needed Asterisk modules res_jabber.so and
>>> >> > chan_gtalk.so are loaded?
>>> >> >
>>> >> > Michael
>>> >>
>>> >>
>>> >> >From what your status tab shows, you are using AstLinux 1.0.5 (latest
>>> >> version) but with Asterisk 1.4.44 (deprecated).
>>> >> You can check your modules in the Asterisk CLI with "module show"
>>> >>
>>> >> Michael
>>> >>
>>> >>
>>> >>
>>> >> ------------------------------------------------------------------------------
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>>> >> Remotely access PCs and mobile devices and provide instant support
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>>> >> _______________________________________________
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>>> >> [email protected]
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>>> >>
>>> >> Donations to support AstLinux are graciously accepted via PayPal to
>>> >> [email protected].
>>> >>
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>>> >> Remotely access PCs and mobile devices and provide instant support
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>>> >> http://p.sf.net/sfu/logmein_12329d2d_______________________________________________
>>> >> Astlinux-users mailing list
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>>> >>
>>> >> Donations to support AstLinux are graciously accepted via PayPal to
>>> >> [email protected].
>>> >
>>> >
>>> >
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>>> > Remotely access PCs and mobile devices and provide instant support
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>>> > [email protected]
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>>> >
>>> > Donations to support AstLinux are graciously accepted via PayPal to
>>> > [email protected].
>>
>>
>>
>> ------------------------------------------------------------------------------
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>> _______________________________________________
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>> [email protected]
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>>
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>> [email protected].
>
>
>
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>
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