Re: [asterisk-users] Set ringtone by dialed number

2013-07-23 Thread Josh Hopkins 
The reason for this is we have one primary company office but there are two 
entities and if someone call the Denver number it would be for 1 organization 
and would ring differently helping our staff remember how to answer the phone 
for the Denver organization rather than for the Colorado Springs number which 
is a different entity.   A lot of times people are rushing to answer the phone 
and do not look at the callerID this would give them and auditory reminder of 
how they need to answer the phone.

How would I go about setting up telling the phone to change the ring tone in 
the SIP header?

On Monday 22 July 2013, Josh Hopkins wrote:
 Would it be possible to set the ringtone based on the number that was
 dialed?
If the phones you are using allow the ringing tone to be changed by sending a
SIP header, yes.

 Example of what the goal is:
 Dial Denver number
 Incoming calls ring with ringtone  1

 Dial main number
 Incoming calls ring with ringtone 2

But why would anyone want this?  What is the point of changing the sound that
your phone makes when someone calls you, depending on who you called last?
 We are currently using Digium D40, D50, D70 phones.


--
AJS

Answers come *after* questions.



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Hopkins
Sent: Monday, July 22, 2013 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set ringtone by dialed number

Would it be possible to set the ringtone based on the number that was dialed?

Example of what the goal is:
Dial Denver number
Incoming calls ring with ringtone  1

Dial main number
Incoming calls ring with ringtone 2

We are currently using Digium D40, D50, D70 phones.
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[asterisk-users] Set ringtone by dialed number

2013-07-22 Thread Josh Hopkins 
Would it be possible to set the ringtone based on the number that was dialed?

Example of what the goal is:
Dial Denver number
Incoming calls ring with ringtone  1

Dial main number
Incoming calls ring with ringtone 2

We are currently using Digium D40, D50, D70 phones.
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[asterisk-users] inboun routing based on area aode

2012-09-17 Thread Josh Hopkins 
I am currently using AsteriskNow v2.  

What I am trying to accomplish is having all calls from an area code go 
directly to the person responsible for that area.  While searching for a 
solution for this I did come across a post that had a few examples.  So Josh at 
extension 1902 would receive all calls from the 808 area code.

exten = s,1,GotoIf($${CALLERIDNUM:0:3} = 808?1902|1)

While asterisknow uses freepbx to control the config files. Where and how would 
I go about putting this into freepbx or another loaded config file that where 
something like the above would work.  Thanks,
/Josh



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[asterisk-users] Call Recording

2012-08-28 Thread Josh Hopkins 

I am trying to record calls on demand both inbound and outbound calls.  I can 
record outbound calls just fine but not inbound calls or calls from an 
internally between extensions.   I am using the latest asterisk 1.8.x certified 
version.
On an outbound call I see:
== Using SIP RTP CoS mark 5
-- Called SIP/ BVTrunk /719000
-- SIP/BVTrunk-0163 is making progress passing it to SIP/1010-0162
-- SIP/BVTrunk-0163 answered SIP/1010-0162
--  Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0162, 
0?Set(THISEXTEN=719)) in new stack
-- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0162, 
1?Set(THISEXTEN=1010)) in new stack
-- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0162, 
0?MacroExit()) in new stack
-- Executing [s@macro-one-touch-record:4] GotoIf(SIP/1010-0162, 
0?stoprec) in new stack
-- Executing [s@macro-one-touch-record:5] GotoIf(SIP/1010-0162, 
0?stopped) in new stack
-- Executing [s@macro-one-touch-record:6] GotoIf(SIP/1010-0162, 
0?recording) in new stack
-- Executing [s@macro-one-touch-record:7] Set(SIP/1010-0162, 
MASTER_CHANNEL(ONETOUCH_REC)=RECORDING) in new stack
-- Executing [s@macro-one-touch-record:8] Set(SIP/1010-0162, 
MASTER_CHANNEL(REC_STATUS)=RECORDING) in new stack
-- Executing [s@macro-one-touch-record:9] Set(SIP/1010-0162, 
AUDIOHOOK_INHERIT(MixMonitor)=yes) in new stack
-- Executing [s@macro-one-touch-record:10] MixMonitor(SIP/1010-0162, 
2012/08/21/out-719000-1010-20120821-183119-1345595479.530.wav,a,) in new 
stack
  == Begin MixMonitor Recording SIP/1010-0162
-- Executing [s@macro-one-touch-record:11] Set(SIP/1010-0162, 
MON_FMT=wav) in new stack
-- Executing [s@macro-one-touch-record:12] Set(SIP/1010-0162, 
MASTER_CHANNEL(CDR(recordingfile))=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack
-- Executing [s@macro-one-touch-record:13] Set(SIP/1010-0162, 
MASTER_CHANNEL(ONETOUCH_RECFILE)=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack
-- Executing [s@macro-one-touch-record:14] Playback(SIP/1010-0162, 
beep) in new stack
-- SIP/1010-0162 Playing 'beep.ulaw' (language 'en')
-- Executing [s@macro-one-touch-record:15] MacroExit(SIP/1010-0162, 
) in new stack
-- Executing [h@macro-dialout-trunk:1] Macro(SIP/1010-0162, 
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/1010-0162, 1?theend) 
in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf(SIP/1010-0162, 
1?Set(CDR(recordingfile)=out-719000-1010-20120821-183119-1345595479.530.wav))
 in new stack
-- Executing [s@macro-hangupcall:4] Hangup(SIP/1010-0162, ) in new 
stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'SIP/1010-0162' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 
'SIP/1010-0162'
  == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 
'SIP/1010-0162' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 719000, 6) exited non-zero on 
'SIP/1010-0162'
  == MixMonitor close filestream
  == End MixMonitor Recording SIP/1010-0162
  == Extension Changed 1010[ext-local] new state Idle for Notify User 1004
On inbound calls I see:
== Using SIP RTP CoS mark 5
-- Called SIP/1010
-- Connected line update to SIP/ BVTrunk -0160 prevented.
  == Extension Changed 1010[ext-local] new state Ringing for Notify User 1004
-- SIP/1010-0161 is ringing
-- Connected line update to SIP/ BVTrunk -0160 prevented.
-- SIP/1010-0161 answered SIP/ BVTrunk -0160
  == Extension Changed 1010[ext-local] new state InUse for Notify User 1004
-- Executing [s@macro-auto-blkvm:1] Set(SIP/1010-0161, 
__MACRO_RESULT=) in new stack
-- Executing [s@macro-auto-blkvm:2] Macro(SIP/1010-0161, 
blkvm-clr,) in new stack
-- Executing [s@macro-blkvm-clr:1] Set(SIP/1010-0161, 
SHARED(BLKVM,SIP/BVTrunk-0160)=) in new stack
-- Executing [s@macro-blkvm-clr:2] Set(SIP/1010-0161, 
GOSUB_RETVAL=) in new stack
-- Executing [s@macro-blkvm-clr:3] MacroExit(SIP/1010-0161, ) in 
new stack
-- Executing [s@macro-auto-blkvm:3] ExecIf(SIP/1010-0161, 
0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=1010)) in new stack
-- Executing [s@macro-auto-blkvm:4] ExecIf(SIP/1010-0161, 
0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Josh Hopkins)) in new stack
--  Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0161, 
0?Set(THISEXTEN=1010)) in new stack
-- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0161, 
1?Set(THISEXTEN=1010)) in new stack
-- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0161, 
1?MacroExit()) in new stack

Re: [asterisk-users] recording calls

2012-08-22 Thread Josh Hopkins 
Being new to asterisk I really don't understand what I am looking at.

[macro-one-touch-record]
include = macro-one-touch-record-custom
exten = 
s,1,ExecIf($[${PICKUP_EXTEN}!=]?Set(THISEXTEN=${CUT(CALLFILENAME,-,2)}))
exten = 
s,n,ExecIf($[${THISEXTEN}=]?Set(THISEXTEN=${IF($[${REALCALLERIDNUM}=]?${CUT(CALLFILENAME,-,2)}:${FROMEXTEN})}))
exten = s,n,ExecIf($[${CUT(CALLFILENAME,-,1)}=exten  
${DB(AMPUSER/${THISEXTEN}/recording/ondemand)}!=enabled]?MacroExit())
exten = s,n,GotoIf($[${MASTER_CHANNEL(ONETOUCH_REC)}=RECORDING]?stoprec)
exten = s,n,GotoIf($[${MASTER_CHANNEL(REC_POLICY_MODE)}=never]?stopped)
exten = s,n,GotoIf($[${MASTER_CHANNEL(ONETOUCH_REC)}=  
${MASTER_CHANNEL(REC_STATUS)}=RECORDING]?recording)
exten = s,n,Set(MASTER_CHANNEL(ONETOUCH_REC)=RECORDING)
exten = s,n,Set(MASTER_CHANNEL(REC_STATUS)=RECORDING)
exten = s,n(mixmon),Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 
s,n,MixMonitor(${MIXMON_DIR}${YEAR}/${MONTH}/${DAY}/${CALLFILENAME}.${MIXMON_FORMAT},a,${MIXMON_POST})
exten = 
s,n,Set(MON_FMT=${IF($[${LEN(${MIXMON_FORMAT})}]?${MIXMON_FORMAT}:wav)})
exten = s,n,Set(MASTER_CHANNEL(CDR(recordingfile))=${CALLFILENAME}.${MON_FMT})
exten = s,n,Set(MASTER_CHANNEL(ONETOUCH_RECFILE)=${CALLFILENAME}.${MON_FMT})
exten = s,n(recording),Playback(beep)
exten = s,n,MacroExit()
exten = s,n(stoprec),StopMixMonitor()
exten = s,n,Set(MASTER_CHANNEL(ONETOUCH_REC)=PAUSED)
exten = s,n,Set(MASTER_CHANNEL(REC_STATUS)=PAUSED)
exten = 
s,n,ExecIf($[${THISEXTEN}=]?Set(THISEXTEN=${IF($[${REALCALLERIDNUM}=]?${DIALEDPEERNUMBER}:${FROMEXTEN})}))
exten = s,n(stopped),Playback(beepbeep)
exten = s,n,MacroExit()

;--== end of [macro-one-touch-record] ==--;

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: Wednesday, August 22, 2012 6:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] recording calls

you need to provide dial plan for macro-one-touch-record.

i think there is something which records outgoing only
On Wed, Aug 22, 2012 at 6:39 AM, Josh Hopkins 
j...@prorivertech.commailto:j...@prorivertech.com wrote:
I am trying to record calls on demand both inbound and outbound calls.  I can 
record outbound calls just fine but not inbound calls or calls from an 
internally between extensions.   I am using the latest asterisk 1.8.x certified 
version.

On an outbound call I see:

== Using SIP RTP CoS mark 5
-- Called SIP/ BVTrunk /719000
-- SIP/BVTrunk-0163 is making progress passing it to SIP/1010-0162
-- SIP/BVTrunk-0163 answered SIP/1010-0162
--  Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0162, 
0?Set(THISEXTEN=719)) in new stack
-- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0162, 
1?Set(THISEXTEN=1010)) in new stack
-- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0162, 
0?MacroExit()) in new stack
-- Executing [s@macro-one-touch-record:4] GotoIf(SIP/1010-0162, 
0?stoprec) in new stack
-- Executing [s@macro-one-touch-record:5] GotoIf(SIP/1010-0162, 
0?stopped) in new stack
-- Executing [s@macro-one-touch-record:6] GotoIf(SIP/1010-0162, 
0?recording) in new stack
-- Executing [s@macro-one-touch-record:7] Set(SIP/1010-0162, 
MASTER_CHANNEL(ONETOUCH_REC)=RECORDING) in new stack
-- Executing [s@macro-one-touch-record:8] Set(SIP/1010-0162, 
MASTER_CHANNEL(REC_STATUS)=RECORDING) in new stack
-- Executing [s@macro-one-touch-record:9] Set(SIP/1010-0162, 
AUDIOHOOK_INHERIT(MixMonitor)=yes) in new stack
-- Executing [s@macro-one-touch-record:10] MixMonitor(SIP/1010-0162, 
2012/08/21/out-719000-1010-20120821-183119-1345595479.530.wav,a,) in new 
stack
  == Begin MixMonitor Recording SIP/1010-0162
-- Executing [s@macro-one-touch-record:11] Set(SIP/1010-0162, 
MON_FMT=wav) in new stack
-- Executing [s@macro-one-touch-record:12] Set(SIP/1010-0162, 
MASTER_CHANNEL(CDR(recordingfile))=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack
-- Executing [s@macro-one-touch-record:13] Set(SIP/1010-0162, 
MASTER_CHANNEL(ONETOUCH_RECFILE)=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack
-- Executing [s@macro-one-touch-record:14] Playback(SIP/1010-0162, 
beep) in new stack
-- SIP/1010-0162 Playing 'beep.ulaw' (language 'en')
-- Executing [s@macro-one-touch-record:15] MacroExit(SIP/1010-0162, 
) in new stack
-- Executing [h@macro-dialout-trunk:1] Macro(SIP/1010-0162, 
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/1010-0162, 1?theend) 
in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf(SIP/1010-0162, 
1?Set(CDR(recordingfile)=out-719000-1010-20120821-183119-1345595479.530.wav))
 in new stack
-- Executing [s@macro-hangupcall:4] Hangup

[asterisk-users] recording calls

2012-08-21 Thread Josh Hopkins 
I am trying to record calls on demand both inbound and outbound calls.  I can 
record outbound calls just fine but not inbound calls or calls from an 
internally between extensions.   I am using the latest asterisk 1.8.x certified 
version.

On an outbound call I see:

== Using SIP RTP CoS mark 5
-- Called SIP/ BVTrunk /719000
-- SIP/BVTrunk-0163 is making progress passing it to SIP/1010-0162
-- SIP/BVTrunk-0163 answered SIP/1010-0162
--  Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0162, 
0?Set(THISEXTEN=719)) in new stack
-- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0162, 
1?Set(THISEXTEN=1010)) in new stack
-- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0162, 
0?MacroExit()) in new stack
-- Executing [s@macro-one-touch-record:4] GotoIf(SIP/1010-0162, 
0?stoprec) in new stack
-- Executing [s@macro-one-touch-record:5] GotoIf(SIP/1010-0162, 
0?stopped) in new stack
-- Executing [s@macro-one-touch-record:6] GotoIf(SIP/1010-0162, 
0?recording) in new stack
-- Executing [s@macro-one-touch-record:7] Set(SIP/1010-0162, 
MASTER_CHANNEL(ONETOUCH_REC)=RECORDING) in new stack
-- Executing [s@macro-one-touch-record:8] Set(SIP/1010-0162, 
MASTER_CHANNEL(REC_STATUS)=RECORDING) in new stack
-- Executing [s@macro-one-touch-record:9] Set(SIP/1010-0162, 
AUDIOHOOK_INHERIT(MixMonitor)=yes) in new stack
-- Executing [s@macro-one-touch-record:10] MixMonitor(SIP/1010-0162, 
2012/08/21/out-719000-1010-20120821-183119-1345595479.530.wav,a,) in new 
stack
  == Begin MixMonitor Recording SIP/1010-0162
-- Executing [s@macro-one-touch-record:11] Set(SIP/1010-0162, 
MON_FMT=wav) in new stack
-- Executing [s@macro-one-touch-record:12] Set(SIP/1010-0162, 
MASTER_CHANNEL(CDR(recordingfile))=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack
-- Executing [s@macro-one-touch-record:13] Set(SIP/1010-0162, 
MASTER_CHANNEL(ONETOUCH_RECFILE)=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack
-- Executing [s@macro-one-touch-record:14] Playback(SIP/1010-0162, 
beep) in new stack
-- SIP/1010-0162 Playing 'beep.ulaw' (language 'en')
-- Executing [s@macro-one-touch-record:15] MacroExit(SIP/1010-0162, 
) in new stack
-- Executing [h@macro-dialout-trunk:1] Macro(SIP/1010-0162, 
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/1010-0162, 1?theend) 
in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf(SIP/1010-0162, 
1?Set(CDR(recordingfile)=out-719000-1010-20120821-183119-1345595479.530.wav))
 in new stack
-- Executing [s@macro-hangupcall:4] Hangup(SIP/1010-0162, ) in new 
stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'SIP/1010-0162' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 
'SIP/1010-0162'
  == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 
'SIP/1010-0162' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 719000, 6) exited non-zero on 
'SIP/1010-0162'
  == MixMonitor close filestream
  == End MixMonitor Recording SIP/1010-0162
  == Extension Changed 1010[ext-local] new state Idle for Notify User 1004

On inbound calls I see:

== Using SIP RTP CoS mark 5
-- Called SIP/1010
-- Connected line update to SIP/ BVTrunk -0160 prevented.
  == Extension Changed 1010[ext-local] new state Ringing for Notify User 1004
-- SIP/1010-0161 is ringing
-- Connected line update to SIP/ BVTrunk -0160 prevented.
-- SIP/1010-0161 answered SIP/ BVTrunk -0160
  == Extension Changed 1010[ext-local] new state InUse for Notify User 1004
-- Executing [s@macro-auto-blkvm:1] Set(SIP/1010-0161, 
__MACRO_RESULT=) in new stack
-- Executing [s@macro-auto-blkvm:2] Macro(SIP/1010-0161, 
blkvm-clr,) in new stack
-- Executing [s@macro-blkvm-clr:1] Set(SIP/1010-0161, 
SHARED(BLKVM,SIP/BVTrunk-0160)=) in new stack
-- Executing [s@macro-blkvm-clr:2] Set(SIP/1010-0161, 
GOSUB_RETVAL=) in new stack
-- Executing [s@macro-blkvm-clr:3] MacroExit(SIP/1010-0161, ) in 
new stack
-- Executing [s@macro-auto-blkvm:3] ExecIf(SIP/1010-0161, 
0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=1010)) in new stack
-- Executing [s@macro-auto-blkvm:4] ExecIf(SIP/1010-0161, 
0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Josh Hopkins)) in new stack
--  Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0161, 
0?Set(THISEXTEN=1010)) in new stack
-- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0161, 
1?Set(THISEXTEN=1010)) in new stack
-- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0161, 
1?MacroExit()) in new stack

[asterisk-users] Digium Phones

2012-08-20 Thread Josh Hopkins 
I have been looking for the specs (format, bit rate, ect) on custom ringtones 
for digium phones.  Using the DPMA how would I deliver the ringtone to a digium 
phone?
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Re: [asterisk-users] Digium Phones

2012-08-20 Thread Josh Hopkins 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Rusty Newton
 Sent: Monday, August 20, 2012 9:23 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Digium Phones
 
 On 8/20/2012 10:14 AM, Josh Hopkins wrote:
 
  I have been looking for the specs (format, bit rate, ect) on custom
  ringtones for digium phones.  Using the DPMA how would I deliver the
  ringtone to a digium phone?
 
 
 https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration
 
 All the answers you seek are located there. Ctrl + f for ringtone
 throughout the document.
 
 Also, 16-bit, 16kHz mono .wav according to the wiki.
 
 --
 Rusty Newton
 Digium, Inc | Open Source Community Support Manager Check us out at:
 www.digium.com www.asterisk.org
 
 
 --
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 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[Josh Hopkins] 
Thanks for the reply.  I do see it now.  Not sure how I missed all that before. 
 One thing I don't think I did see was where I should place my ringtone file?  
I am guessing that if I put it in the same location as our custom imagefile 
that it might work.  Is there a url_prefix option like there is for firmware? 


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[asterisk-users] Voicemail Emails

2012-07-20 Thread Josh Hopkins 
Has anyone been able to make an html template for the voicemail emails. We 
would love to customize them beyond just plain text. We have dome some Google 
searches and have not been able to come up with much.

I believe that Switchvox has customized the voicemail email  into html.  Has 
anyone ever tried this?  Thanks,
/Josh

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Re: [asterisk-users] Voicemail Emails

2012-07-20 Thread Josh Hopkins 
Yes and you just get html code in the email rather than the html format.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi
Sent: Friday, July 20, 2012 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Emails

Have you tried to insert the HTML code directly into the body?

Il 20/07/12 19:53, Josh Hopkins ha scritto:
Has anyone been able to make an html template for the voicemail emails. We 
would love to customize them beyond just plain text. We have dome some Google 
searches and have not been able to come up with much.

I believe that Switchvox has customized the voicemail email  into html.  Has 
anyone ever tried this?  Thanks,
/Josh
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