Re: [asterisk-users] OT: Grandstream, call pickup, ...
I got the call pick up to work with the Digium AA50 and the GXP2000.
Here is what I used in the dial plan:
exten=_*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten=_*8.,n,Pickup(${EXTEN:2...@pickupmark)
You dial *8 and the extension that is ringing and you will intercept the call.
Hope that helps.
Before printing this e-mail think if it is necessary. Think Green!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, June 09, 2009 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Grandstream, call pickup, ...
On Tue, 9 Jun 2009, Steve Repo wrote:
On Tue, Jun 9, 2009 at 4:50 PM, Philipp
Kempgenphilipp.kemp...@amooma.de wrote:
Thomas Kenyon schrieb:
Peder wrote:
Decent product, but their support and development are horrible. I
showed them that their SIP over TCP implementation was broken and
their reply was use udp
Such a shame it sounds like it has gone down hill, previously when
I've spoken to them the standard response was that they'll pass my
comments on to the development team.
The development team a.k.a. /dev/null? ;-)
Pretty much. I have a GXP 1200 which sorta works ok. I hope they will
work on their products and customer service.
They do have a couple of firmwares in beta which might make things better?
Steve
I now have a handful of the new 3140 video phones, and I have to admit I am
pretty impressed. I have a handful of the 3000 phones, and was somewhat
impressed. We are planning to offer a service around the 3140 soon, and I am
trudging through the provisioning...
j
___
-- 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
Re: [asterisk-users] intercom/paging with grandstream gxp2000
Thanks for your reply!
Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with Family, however, the intercom
configuration that I am currently testing makes all my calls and intercom
call. It does not matter if I call using Dial or Page on the GXP2000, the
call is always and intercom call. For some reason the GXP2000 is receiving
the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
the configuration by looking at the configuration below?
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
what would you do differently?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Thursday, August 07, 2008 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000
On Wed, 6 Aug 2008, Fidel Garcia wrote:
Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom. I know I
must be doing something wrong with the way I am adding the steps to my
call;
I am not familiar with variables and flags.
What *exactly* are you trying to achieve?
I have used both paging and intercom mode in the Grandstreams with good
results.
You do need the settings in the phone set ON - ie.
Allow Auto Answer by Call-Info: No Yes
Turn off speaker on remote disconnect: No Yes
These both need to be set to YES or ON.
That won't affect normal calls to that account on the phone - although the
turn off speaker one does make the phone easier to use IMO...
So call the phone and the person answers normally, as before, but if you
rhen add the SIP header:
SIPAddHeader(Call-Info: answer-after=0)
The phone will auto-answer - when the next Dial or Page command is
directed to it.
What next? If you want to Page the phone, use the Page() application.
So if the phone is SIP/100 then to Dial the phone normally..
exten = 100,1,Dial(SIP/100)
but to page it:
exten = 200,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 200,n,Page(SIP/100)
and to intercom to it:
exten = 300,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 300,n,Page(SIP/100,d)
So this has added 3 new extensions, 100, 200 and 300 - which all 'call'
SIP/100, but in 3 differet ways.
Gordon
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.5.12/1596 - Release Date: 8/6/2008
4:55 PM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intercom/paging with grandstream gxp2000
I added the configuration as you suggest but now the phone does not do
intercom. I tried Dial and Page in the gxp2000 but everything goes out as
Dial.
Here is the extensions.conf now
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?3:4)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
Any idea? I am very bad on this asterisk thing, sorry guys.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Thursday, August 07, 2008 12:10 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000
On 10:59, Thu 07 Aug 08, Fidel Garcia wrote:
Thanks for your reply!
Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with Family, however, the intercom
configuration that I am currently testing makes all my calls and intercom
call. It does not matter if I call using Dial or Page on the GXP2000, the
call is always and intercom call. For some reason the GXP2000 is receiving
the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
the configuration by looking at the configuration below?
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
if the sip header Call-Info has value answer-after=0 it goes to prio 2,
otherwise 3
Now let's have a closer look at those.
Hhmm, prio two is the gotoif, prio three adds the answer-after=0 ...
I think you mean:
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?3:4)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
what would you do differently?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Thursday, August 07, 2008 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000
On Wed, 6 Aug 2008, Fidel Garcia wrote:
Guys I have been reading for days on how to get this to work with
asterisk
and for some reason every time I call the call goes to intercom. I know
I
must be doing something wrong with the way I am adding the steps to my
call;
I am not familiar with variables and flags.
What *exactly* are you trying to achieve?
I have used both paging and intercom mode in the Grandstreams with good
results.
You do need the settings in the phone set ON - ie.
Allow Auto Answer by Call-Info: No Yes
Turn off speaker on remote disconnect: No Yes
These both need to be set to YES or ON.
That won't affect normal calls to that account on the phone - although the
turn off speaker one does make the phone easier to use IMO...
So call the phone and the person answers normally, as before, but if you
rhen add the SIP header:
SIPAddHeader(Call-Info: answer-after=0)
The phone will auto-answer - when the next Dial or Page command is
directed to it.
What next? If you want to Page the phone, use the Page() application.
So if the phone is SIP/100 then to Dial the phone normally..
exten = 100,1,Dial(SIP/100)
but to page it:
exten = 200,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 200,n,Page(SIP/100)
and to intercom to it:
exten = 300,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 300,n,Page(SIP/100,d)
So this has added 3 new extensions, 100, 200 and 300 - which all 'call'
SIP/100, but in 3 differet ways.
Gordon
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.5.12/1596 - Release Date: 8/6/2008
4:55 PM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit
[asterisk-users] intercom/paging with grandstream gxp2000
Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom. I know I
must be doing something wrong with the way I am adding the steps to my call;
I am not familiar with variables and flags.
Here is my configuration:
Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware.
Extensions.conf:
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)
exten=s,2,SIPAddHeader(Call-Info: answer-after=0)
exten=s,3,Dial(${ARG2},20)
exten=s,4,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
GXP2000 configuration:
Under Account1 I checked options:
Allow Auto Answer by Call-Info: No Yes
Turn off speaker on
remote disconnect: No Yes
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax: (305)-477-0013
http://www.systeamusa.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intercom/paging with grandstream gxp2000
I am sorry, this is the actual extensions.conf:
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
As you can see here Goto and SIPAddHeader are 2 and 3. In the prior email I
had both lines under 2.
Fidel Garcia
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia
Sent: Wednesday, August 06, 2008 5:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] intercom/paging with grandstream gxp2000
Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom. I know I
must be doing something wrong with the way I am adding the steps to my call;
I am not familiar with variables and flags.
Here is my configuration:
Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware.
Extensions.conf:
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)
exten=s,2,SIPAddHeader(Call-Info: answer-after=0)
exten=s,3,Dial(${ARG2},20)
exten=s,4,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
GXP2000 configuration:
Under Account1 I checked options:
Allow Auto Answer by Call-Info: No Yes
Turn off speaker on
remote disconnect: No Yes
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax: (305)-477-0013
http://www.systeamusa.com
No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.5.12/1595 - Release Date: 8/6/2008
8:23 AM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line 0005 cannot be answered?
I have a Digium Appliance AA50 configure with 8 lines and two dial plans. Each dial plan takes care of a particular location. In Dialplan2 we have 4 lines. xxx xxx 0333 xxx xxx 0005 xxx xxx 0006 xxx xxx 0007 When a call gets to line 0005 you pick up the phone but the call does not get connected, you can still hear the phone ringing in a noisy weird way on the handset - the caller's phone never stops ringing. Is this a problem related to the way the line was taken out of the punch panel or does I have to do anything with configuration? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] custom configuration with appliance aa50.
I have just received a list of requests from one of our customers and I really do not have the time or knowledge to work on it. I will truly appreciate it if someone could help me outside of the mailing list. Please contact me if interested. Digium Appliance AA50/GrandStream GXP2000. Here is the list: 1. We need to be able to program the buttons along the right side of the phone so we can see who is on the phone. 2. How to implement phone paging/intercom. 3. If the front desk transfers a call to Jerry's phone but he isn't there how can I grab that call from my phone so it doesn't go to his voicemail? 4. If we transfer a call can we send it directly to voice mail or does it have to ring on the person's phone first? 5. How do we set after hour message and holiday messages? 6. Can we set a different ring tones for incoming calls and ext. to ext calls? 7. Is there an intercom on the phone for ext to ext. calls or do we have to pick up the handset to hear the person calling you? 8. When a call is transferred to another extension provide a recording that the caller can hear: Leave a message or press zero to return to the operator. If ZERO is entered can we go back to the extension the call came from? Instead of dropping it to the pool again. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] increase ring time out
Thank you so much! It works perfectly fine!
I went to extensions.conf and edited stdexten macro to 30.
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax: (305)-477-0013
http://www.systeamusa.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Bailey
Sent: Thursday, July 24, 2008 6:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] increase ring time out
- Fidel Garcia [EMAIL PROTECTED] wrote:
Where exactly do I have to change it?
The GUI on the AA50 generates users via users.conf. These users are added
into
the dialplan automatically and are placed into the default context. Calls
to
the users are made via the stdexten macro. In that macro is a Dial
statement
with a timeout of 20. You would have to adjust that timeout manually and
save
it off (Run the save_config script)
One caveat is that the AA50 is not supported when you manually modify the
dial
plan. The changes you make are at your own risk.
- Doug Bailey
This is the extensions.conf file:
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Tue Jul 22 15:14:28 2008
;!
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
trunk_1 = Zap/g1
trunk_1_cid = asreceived
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include = dundi-e164-canonical
include = dundi-e164-customers
include = dundi-e164-via-pstn
[dundi-e164-switch]
switch = DUNDi/e164
[dundi-e164-lookup]
include = dundi-e164-local
include = dundi-e164-switch
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164-lookup
[macro-trunkdial]
exten = s,1,set(CALLERID(all)=${IF(${LEN(${CALLERID(num)})} 6 ?
${CALLERID(al
l)} : ${ARG2})})
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
[iaxtel700]
exten =
_91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider]
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
ignorepat = 9
include = local
include = trunkld
[local]
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten]
exten = s,1,Dial(${ARG2},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-stdPrivacyexten]
exten = s,1,Dial(${ARG2},20|p)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = s-DONTCALL,1,Goto(${ARG3},s,1)
exten = s-TORTURE,1,Goto(${ARG4},s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-page]
exten = s,1,ChanIsAvail(${ARG1}|js)
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
exten = s,n(autoanswer),Set(_ALERT_INFO=RA)
exten = s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten = s,n,NoOp()
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
[demo]
exten = s,1,Wait(1)
exten = s,n,Answer
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n(restart),BackGround(demo-congrats)
exten = s,n(instruct),BackGround(demo-instruct)
exten = s,n,WaitExten
exten = 2,1,BackGround(demo-moreinfo)
exten = 2,n,Goto(s,instruct)
exten = 3,1,Set(LANGUAGE()=fr)
exten = 3,n,Goto(s,restart)
exten = 1000,1,Goto(default,s,1)
exten = 1234,1,Playback(transfer,skip)
exten = 1234,n,Macro(stdexten,1234,${CONSOLE})
exten = 1235,1,Voicemail(u1234)
exten = 1236,1,Dial(Console/dsp)
exten = 1236,n,Voicemail(u1234)
exten = #,1,Playback(demo-thanks)
exten = #,n,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
exten = 500,1,Playback(demo-abouttotry)
exten = 500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten
Re: [asterisk-users] increase ring time out
,${}/${EXTEN:4})
comment = _9256XXX!,1,Local,standard
exten = _9011XXX!,1,Macro(trunkdial,${}/${EXTEN:1})
comment = _9011XXX!,1,International,standard
exten = _9XXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid})
comment = _9XXX!,1,Local,standard
exten =
_91XX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid})
comment = _91XX!,1,Longdistance,standard
exten = _911!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
comment = _911!,1,911,standard
[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
hasbeensetup = Y
[DID_trunk_1]
include = default
exten = _X.,1,Goto(ringroups-custom-1,s,1)
exten = s,1,ExecIf($[ ${CALLERID(num)}= ],SetCallerPres,unavailable)
exten = s,2,ExecIf($[ ${CALLERID(num)}= ],Set,CALLERID(all)=unknown
000 )
exten = s,3,Goto(ringroups-custom-1,s,1)
[ringroups-custom-1]
gui_ring_groupname = Ring All
exten = s,1,NoOp(RINGGROUP)
exten = s,n,Dial(SIP/101SIP/102SIP/103SIP/104SIP/902SIP/903,20,i)
exten = s,n,Voicemail(101,b)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vazquez David
Sent: Tuesday, July 22, 2008 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] increase ring time out
Fidel Garcia wrote:
I need to increase the ringing timeout on the AA50 appliance. How do I
accomplish this?
I need the phones to ring a bit more before the caller gets to the
voicemail.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Could you show your extensions.conf? Normally you'd do that in the Dial
command:
exten = _XX,1,Answer
exten = _XX,n,Dial(SIP/1,20)
...
Where 20 is the time you're letting the phone ring... :-)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.5.4/1566 - Release Date: 7/22/2008
6:00 AM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] increase ring time out
I need to increase the ringing timeout on the AA50 appliance. How do I accomplish this? I need the phones to ring a bit more before the caller gets to the voicemail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] distintive ring
Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distintive ring
This one! The sound of a phone that signals a call coming from internal/external My phones are SIP, I do not know what ZAP means or what it does. Thanks for your reply! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 15, 2008 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] distintive ring Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.4.11/1553 - Release Date: 7/15/2008 5:48 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tone differentiation
I have two different scenario where I would like to apply different tones: 1. Incoming calls should have a different ringing tone than transfer calls 2. While on a call, transfer calls should have a different beep sound on the handset than incoming calls. How/where can I accomplish this? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to change http port on appliance?
I have the AA50 configured with a public address and I would like to change the default http port (80) to something else for security reasons. I cannot find the httpd.conf file anywhere. Help! Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change http port on appliance?
I just found it at : /ramfs/etc/asterisk/http.conf How do I restart the http service without affecting the phone service? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia Sent: Wednesday, July 02, 2008 4:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to change http port on appliance? I have the AA50 configured with a public address and I would like to change the default http port (80) to something else for security reasons. I cannot find the httpd.conf file anywhere. Help! Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com No virus found in this incoming message. Checked by AVG. Version: 8.0.134 / Virus Database: 270.4.3/1529 - Release Date: 7/1/2008 7:23 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change http port on appliance?
Figured it out: Modified the port number at : /ramfs/etc/asterisk/http.conf I accessed the web management and hit Apply Changes. Works fine! Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia Sent: Wednesday, July 02, 2008 4:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to change http port on appliance? I just found it at : /ramfs/etc/asterisk/http.conf How do I restart the http service without affecting the phone service? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia Sent: Wednesday, July 02, 2008 4:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to change http port on appliance? I have the AA50 configured with a public address and I would like to change the default http port (80) to something else for security reasons. I cannot find the httpd.conf file anywhere. Help! Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com No virus found in this incoming message. Checked by AVG. Version: 8.0.134 / Virus Database: 270.4.3/1529 - Release Date: 7/1/2008 7:23 PM No virus found in this incoming message. Checked by AVG. Version: 8.0.134 / Virus Database: 270.4.3/1529 - Release Date: 7/1/2008 7:23 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tone Differentiation
I have two different scenario where I would like to apply different tones: 1. Incoming calls should have a different ringing tone than transfer calls 2. While on a call, transfer calls should have a different beep sound on the handset than incoming calls. How/where can I accomplish this? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change http port on appliance?
Well, most dangerous guys out there (noobs) scan port 80 to begin their attacks. As you may know port numbers go from 1 to 65535 and scanning all of them takes a while. I am using 65531 on my box just to stay away from ip ranges scans on default ports. Not big deal, but I feel safer. Next I will change the ssh service to a different port. Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, July 02, 2008 5:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change http port on appliance? On Wednesday 02 July 2008 14:59:39 Fidel Garcia wrote: I have the AA50 configured with a public address and I would like to change the default http port (80) to something else for security reasons. Because, like, changing the http port is uber secure. Or something. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 8.0.134 / Virus Database: 270.4.3/1529 - Release Date: 7/1/2008 7:23 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gxp2000 time.
I am running Asterisk the appliance with GXP2000 telephones. For some reason I cannot get the telephones to update their time automatically. Steps I have taken to solve problem: - Configured static IP address and DNS - Change NTP server twice. First time to Asterisk server and the second time to another NTP server. Computers in the same network are able to access the internet. The telephones should be able to access the internet too. So far no luck, any ideas? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Do not update to Firefox 3, yet?
Yesterday I was installing a brand new appliance box and configuring it using the newest version of Firefox 3; to my surprise, Firefox no longer works with Asterisk web interface. - When I tried to add a new User it will not show the Dial Plan. - When I tried to edit one of the current users I was not able to select one user at the time. Firefox was selecting all the users every time I clicked on top of an User. Anyone else having this problem? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FOLLOWME Vs QUEUE
Try under incoming calls rules to transfer to the queues using Extesion # or Queue name. Try both options and reboot the machine or appliance! Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tariq .. Sent: Friday, June 27, 2008 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FOLLOWME Vs QUEUE Greetings.. i'm having problems when a queue uses RINGALL stratigy .. so i am thinking of FOLLOWME to do the work .. now my question which is best to use?? i have 5 groups of 10 users each.. I use Asterisk 1.4 regards Tarek _ Need to know now? Get instant answers with Windows Live Messenger. IM on your terms. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_W L_Refresh_messenger_062008 No virus found in this incoming message. Checked by AVG. Version: 8.0.101 / Virus Database: 270.4.1/1522 - Release Date: 6/27/2008 8:27 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Do not update to Firefox 3, yet?
Great info! Thanks! However, they do not mention the fact that when you create a new user you cannot select the DialPlan. I wonder if the path fixes both issues. Any idea? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, June 27, 2008 5:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Do not update to Firefox 3, yet? On Fri, Jun 27, 2008 at 10:02:11AM -0400, Fidel Garcia wrote: Yesterday I was installing a brand new appliance box and configuring it using the newest version of Firefox 3; to my surprise, Firefox no longer works with Asterisk web interface. http://bugs.digium.com/12533 The patch there is a patch to javascript files under /var/lib/asterisk/static-http . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 8.0.101 / Virus Database: 270.4.1/1522 - Release Date: 6/27/2008 8:27 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and remote phone.
Hi everyone! I have been reading for a couple of days online in order to setup a remote phone, but no luck so far. The remote phone registers and I can call extensions, but the call only lasts 19-21 seconds before it drops (when I call from remote phone). When someone calls the remote phone we cannot hear each other. The scenario is de following: Office -Router configuration: -Ports being forwarded to Asterisk -UDP = 5060 -UDP = 5004 -UDP = 5060-5065 (configured range on rpc.conf first) House -Westell Modem configured as IP-Passthrough. In other words, the phone has a static public IP address. Phone: -STUN Server configured. I also had to configured symmetrical RTP because on the other side they couldn't hear me. The call only lasts 18-21 seconds until it drops. Asterisk Configuration Files: -I configured external and lan IP. -Enabled NAT for the remote extension. What am I missing? Why is it that I can call from remote phone and hear the person, but they cannot hear me. When the office calls the house we cannot hear each other. Today I configured Asterisk with a public IP address and the home phone work just fine. However, I do think think this is the best setup because having Asterisk expose to the internet is not a good idea. I want to be able to have remote phones with Asterisk behind a router (private IP address). Thanks in advance. Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice only works from one way.
I was never able to get it to work that way. When I had Asterisk in NAT I
was able to make calls, but most of the times they were one way voice.
I was able to get two-way voice when I configured the remote phone using
STUN and Symetrical RTP. However, the calls dropped every 19-20 seconds. I
read several threads online, but nobody explained the requirements in
details. Everything works fine if you have a public IP address or DMZ on
Asterisk.
Good luck and please let me know if you get it up and running.
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax: (305)-477-0013
http://www.systeamusa.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Friday, June 20, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice only works from one way.
Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, Sam Tam [EMAIL PROTECTED] wrote:
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
extensions.conf
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 2003,1,Dial(SIP/2003)
exten = 6000,1,MeetMe(600,i,54321)
;include = lan-phones
[bogon-calls]
exten = _.,1,Congestion
[pstn-incoming]
include = lan-phones
[local-phones]
include = lan-phones
include = pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped are then routed out to the
PSTN
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten = _9,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of Cisco
gateway
;exten = _9,2,Congestion
exten = _9.,2,Congestion
[lan-phones]
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Answer(SIP/1001)
exten = 1001,102,Voicemail(b1001)
exten = 1001,103,Hangup
Cisco 2611 config
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
