Re: [asterisk-users] Configuring Softphone

[Jeroen Eeuwes]

Hi Gary,

> host = dynamic
> defaultip = dynamic

The defaultip line says to Asterisk that this phone is found at the
IP-address associated with the hostname "dynamic". I'm pretty sure
that if you try a command like "ping dynamic" on your command line it
will not return the IP address of your phone.

So you have to either put the real IP address or a resolvable hostname
on the defaultip line OR remove the defaultip line. The latter is
probably the easiest.

Best regards,
Jeroen Eeuwes

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[asterisk-users] Configuring Softphone

[Gary Kuznitz]

I have no idea the correct way to configure this software phone.

It's called Express Talk
The Asterisk box is at IP = WanLocation
Software phone is at IP = WanSoftware
They are not on the same LAN.

What I have in Extensions.conf is:
[gary-incomming]
exten => 1001,1,Dial(IAX2/gogh)
exten => 1001,2,HangUp()
exten => 120,1,Dial(SIP/Gary)
exten => Gary,1,Goto(120,1)
exten => i,1,Playback(invalid)
exten => i,2,Goto(s,1)
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,NoOp(${CALLERID})
exten => s,4,NoOp(${CALLERIDNUM})
exten => s,5,NoOp(${CALLERIDNAME})
exten => s,6,Wait(4)
exten => s,7,Playback(vm-goodbye)
exten => s,8,Wait(2)
exten => s,9,HangUp() 

What I have in Sip.conf is:
[authentication]

[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=no
nat=yes 

[Gary]
type = friend
username = Gary
callerid = 120
secret = 5351
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all  

When I reload the dialplan I get an error from Asterisk saying:
[Dec  7 22:01:48] NOTICE[5630]: chan_sip.c:15593 handle_request_register: 
Registration from '' failed for 'WanSoftware' - No 
matching 
peer found

The Softphone SipTrace log says:
17:25:35 UDP Packet Received from WanLocation:5060 
<<<
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 
192.168.168.7:5060;branch=z9hG4bK03856;received=WanSoftware;rport=16699
From: ;tag=1424
To: ;tag=as214040c6
Call-ID: 1291771532-3856-gar...@localip
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Any ideas on how to configure it better are welcome.

Thank you,

Gary

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[asterisk-users] Configuring Softphone

[Gary Kuznitz]

Hi,

I'm trying to get a softphone configured.  In Sip.conf [general] I found an 
example 
that said I need:
nat=yes
localnet=192.168.xxx.xxx

Is localnet supposed to be a LAN IP or a WAN IP?

Thank you,

Gary

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Re: [asterisk-users] Dahdi issue with Asterisk 1.8.0

[Timothy Legge]

On Tue, Dec 7, 2010 at 8:17 PM, Thomas Perron wrote:

> Do you have any issues with getting audio to bridge?
> I am using 1.8 also.
>

Not so far, but I am still pretty excited to have a dial tone ;-)  Two hours
last night (and a 1:30 am bed time) lost because I missed one line in a
config file.

So far I have only tested dialing out through the Dahdi interface from a
connected analog phone and a sip phone.  That works, but I need to do
something about the echo on the sip phone.

Tim
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Re: [asterisk-users] debug audio or channel

[Steve Edwards]

On Tue, 7 Dec 2010, Thomas Perron wrote:

> A calls B.
> B rings
> Says connected.
> But the call is not bridged and therefor no audio passes.
> very simple dial plan.

> Does anyone have any short answers on how I can fix this problem:

0) Don't use clouds.

1) Start poking about with wireshark and see if audio is flowing.

If the call is connecting, it sounds like 5060 is being passed OK. If 
audio is not being passed, it sounds like... Well, it doesn't sound like 
anything, so the problem may be a router or firewall issue.

-- 
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Dahdi issue with Asterisk 1.8.0

[Thomas Perron]

Do you have any issues with getting audio to bridge?
I am using 1.8 also.


On Tue, Dec 7, 2010 at 12:38 PM, Timothy Legge  wrote:
> Hi
>
> I was using the delivered Ubuntu 1.6.x packages but I wanted to look at
> gtalk integration so I downloaded, compiled and installed the source (after
> removing the Ubuntu packages) have installed the following:
>
> asterisk-1.8.0
> dahdi-linux-complete-2.4.0+2.4.0
> libpri-1.4.11.5
>
> I copied my config back into place and most seems to work, but I cannot get
> my phone that is plugged into the Wildcard TDM400P REV E/F card that I have
> to work.
>
> Basically, I don't hear the dial tone and Asterisk does not register off
> hook events.  I have spent time reviewing my config but I don't see what the
> issue is.
>
> Is there anything I am missing, or can you suggest some additional things to
> look at?
>
> Tim
>
> chan_dahdi.conf
> grep -v "^;" /etc/asterisk/chan_dahdi.conf | grep -v "^$"
>
> [trunkgroups]
> [channels]
> language=en
> context=phones
> signalling=fxo_ks
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=no
> echocancelwhenbridged=no
> group=1
> callgroup=1
> pickupgroup=1
>
> dahdi-channels.conf:
>
> ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
> ;;; line="1 WCTDM/4/0 FXOKS"
> signalling=fxo_ks
> callerid="Channel 1" <4001>
> mailbox=4001
> group=5
> context=phones
> channel => 1
> callerid=
> mailbox=
> group=
> context=default
>
> ;;; line="2 WCTDM/4/1 FXSKS"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=incoming-local
> channel => 2
> callerid=
> group=
> context=default
>
>
>
>
>
>
>
>
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[asterisk-users] debug audio or channel

[Thomas Perron]

Does anyone have any short answers on how I can fix this problem:

A calls B.
B rings
Says connected.
But the call is not bridged and therefor no audio passes.
very simple dial plan.

Frustrated.
v 1.8

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Re: [asterisk-users] 'Bookmarking' a place in a sound file

[David Cunningham]

Steve, that looks just the job, thank you very much.


On Wed, Dec 8, 2010 at 2:32 AM, Steve Edwards wrote:

> On Tue, 7 Dec 2010, David Cunningham wrote:
>
> > Is it possible to somehow 'bookmark' a place in a sound file? That is,
> > the user presses a key while a sound file is playing and that point is
> > saved, and some time in the future we can play the same sound file and
> > tell it to start playing from that point.
> >
> > This would be done within a perl AGI program.
>
> The AGI command 'stream file' will return 'endpos' when interrupted with a
> keypress. You could then save that in a channel variable or a database.
>
> A subsequent call to 'stream file' would include 'endpos' as the 'sample
> offset.'
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] Dahdi issue with Asterisk 1.8.0

[Timothy Legge]

Hi

On Tue, Dec 7, 2010 at 2:57 PM, John Novack
wrote:

>
>>  I have no experience with 1.8, but unless things have changed  channel=
> has to be the last line in a section. the remaining lines are ignored
> Don't you also need [line1] at the beginning of each section?
>
> using context=default has been a security issue in the past. Using a
> different context, and having the default context point to nothing more than
> a rude recording may save you in the case of a security breach
>
> John Novack
>
>
Weird, all the configs seem to be generated with those extra lines.  My
default current talks about weasels, which is causing me issues with gtalk,
but that is another issue.

I was missing:

#include dahdi-channels.conf

from chan_dahdi.conf.

Thanks

Tim
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Re: [asterisk-users] Issue with MOH - Asterisk 1.4.17

[Napoleón Ernesto López Espinoza]

On Tue, Dec 7, 2010 at 9:19 AM, Napoleón Ernesto López Espinoza <
napoleon.lo...@gmail.com> wrote:
Thanks Doug and Darrick for your input. Reloaded again and MOH is working. I
think updating to version 1.4.37 fixed it.
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Re: [asterisk-users] Dahdi issue with Asterisk 1.8.0

[John Novack]

Timothy Legge wrote:
> Hi
>
> I was using the delivered Ubuntu 1.6.x packages but I wanted to look 
> at gtalk integration so I downloaded, compiled and installed the 
> source (after removing the Ubuntu packages) have installed the following:
>
> asterisk-1.8.0
> dahdi-linux-complete-2.4.0+2.4.0
> libpri-1.4.11.5
>
> I copied my config back into place and most seems to work, but I 
> cannot get my phone that is plugged into the Wildcard TDM400P REV E/F 
> card that I have to work.
>
> Basically, I don't hear the dial tone and Asterisk does not register 
> off hook events.  I have spent time reviewing my config but I don't 
> see what the issue is.
>
> Is there anything I am missing, or can you suggest some additional 
> things to look at?
>
> Tim
>
> chan_dahdi.conf
> grep -v "^;" /etc/asterisk/chan_dahdi.conf | grep -v "^$"
>
> [trunkgroups]
> [channels]
> language=en
> context=phones
> signalling=fxo_ks
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=no
> echocancelwhenbridged=no
> group=1
> callgroup=1
> pickupgroup=1
>
> dahdi-channels.conf:
>
> ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
> ;;; line="1 WCTDM/4/0 FXOKS"
> signalling=fxo_ks
> callerid="Channel 1" <4001>
> mailbox=4001
> group=5
> context=phones
> channel => 1
> callerid=
> mailbox=
> group=
> context=default
>
> ;;; line="2 WCTDM/4/1 FXSKS"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=incoming-local
> channel => 2
> callerid=
> group=
> context=default
>
I have no experience with 1.8, but unless things have changed  channel= 
has to be the last line in a section. the remaining lines are ignored
Don't you also need [line1] at the beginning of each section?

using context=default has been a security issue in the past. Using a 
different context, and having the default context point to nothing more 
than a rude recording may save you in the case of a security breach

John Novack

-- 

Dog is my Co-pilot


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Re: [asterisk-users] [headset/mic] Volume too low + echo in *

[Dave Platt]

>   I'm having the following problem when using a headset on XP
> connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus
> motherboard:
> 
> - Using any sound recorder (Windows', Audacity, XLite), the level is
> just too low when speaking at a conversational level, even with the
> microphone level pumped all the way up (line displayed totally flat in
> Recorder)
> http://img704.imageshack.us/img704/7981/headsetlowvolumeecho.jpg
> 
> - In addition, when making a call with XLite and Asterisk, I get a bit
> of echo
> 
> - Same issues when trying with a different headset

Same headset model, or different headset model?

> 
> - Enabling "Auto gain control AGC" in XLite makes no difference.
> 
> Any idea what can be done? Should I use a different soundcard?
> Amplified headset? 

Most computer mic inputs these days, are designed to work with
mics and headsets using electret microphone elements.  These
microphone elements normally have a built-in FET buffer/amplifier,
and have a respectably high audio output level.  The FET amplifier
requires some DC power to operate;  this is normally provided by
the sound card, as a resistor-coupled DC voltage applied to the
mic input pin inside the jack (it's usually in the 3-9 volt range).

Some headsets use "dynamic" (electromagnetic) microphones...
essentially little loudspeakers operated in reverse.  These do not
require DC power from the sound card to operate, but they have a
significantly lower audio output level.  They do require
amplification in order to drive an input designed for electret
microphones.

Some sound cards have mic inputs which are switchable... the
DC power feature can be enabled or disabled, and there's a
"gain boost" setting which switches in a preamplifier stage
(often around 10-20 dB) for use with a dynamic mic.

You may be attempting to use a headset with a dynamic mic,
with a sound card whose mic input was intended only for use with
electret mics and doesn't have a preamplifier.  If this is the
case, switching to a headset with an electret mic and its built-in
FET buffer-amp would probably be your easiest solution.  If that's
what you mean when you refer to an "amplified headset", then yes,
that's probably what you should do.  You wouldn't need a headset
with a separate amplifier-box... the FET amplifier is usually
build right into the microphone element.

It's also possible you have a bad PC sound interface... try
using a different PC with the same headset(s) and see if the
problem persists.

You can probably buy or build a preamp for your existing headset
(I recently built one for a similar purpose) but considering
how inexpensive "A/V" comm/gaming headsets are these days
it's certainly cheaper to buy one.

Another option would be to buy a USB-connected headset...
these have all of the necessary gain electronics in them,
as well as a "USB sound card" chip.  There's only one plug
to plug in, and (once the necessary USB sound drivers are
installed) you could be confident of having the same sound
level and quality on any PC into which you plug it.

>Can something be done in Asterisk about the echo?

How quickly after you speak, do you hear the echo?  Is it
near-instantaneous, or significantly delayed?  What's your
outgoing voice connection (SIP, IAX, or an actual hardwired
phone line with some sort of terminal adapter)?

Does the caller at the far end hear an echo from what s/he
says?  Or does the echo affect only you?



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Re: [asterisk-users] No MOH with parked call

[Steve Davies]

On 7 December 2010 15:00, Steve Davies  wrote:
> On 7 December 2010 14:17, Lee Archer  wrote:
>> Hi, try unloading res_timing_dahdi.so then trying again.
>>
>> Lee
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
>> Davies
>> Sent: 07 December 2010 12:54
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] No MOH with parked call
>>
>> Hi,
>>
>> Has anybody else noticed that MOH does not play on parked calls in
>> 1.6.2.14? Or is it just my setup? MOH seems to work in every other
>> respect (Call Held or in-Queue), but when a call is parked, the logs
>> show MOH being started, but the parked party hears nothing.
>>
>> The verbose logs show the following. Any thoughts on whet to check next?
>>
>> Thanks,
>> Steve
>>
>>
>> ### Call comes in here and is answered
>>    -- SIP/snom360-0d6f answered DAHDI/2-1
>>    -- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f",
>> "0?done") in new stack
>>    -- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f",
>> "CHANNEL(musicclass)=m-default") in new stack
>>    -- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f",
>> "") in new stack
>>
>> ### Here the call is being blind transferred to the Park number
>>    -- Started music on hold, class 'default', on DAHDI/2-1
>>    -- Stopped music on hold on DAHDI/2-1
>>  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
>>    -- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack
>>    -- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in
>> new stack
>>
>> ### Not sure why I send "Ringing" here, but I tried NoOP() and
>> Answer() too just in case
>>    -- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack
>>    -- Executing [...@local:4] Set("DAHDI/2-1",
>> "CHANNEL(musicclass)=default") in new stack
>>    -- Executing [...@local:5] Set("DAHDI/2-1",
>> "CHANNEL(parkinglot)=default") in new stack
>>    -- Executing [...@local:6] Goto("DAHDI/2-1",
>> "parkedcalls_default,park,1") in new stack
>>    -- Goto (parkedcalls_default,park,1)
>>    -- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in
>> new stack
>>  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
>> extension [parkedcalls_default] s, 1 in 90 seconds
>>    -- Added extension '211' priority 1 to parkedcalls_default
>> (0xbe2e528)
>>
>> # The "211" announcement is heard perfectly
>>    --  Playing 'digits/2.alaw' (language 'en')
>>  == Extension Changed 211[extensions] new state InUse for Notify User
>> steve
>>    --  Playing 'digits/1.alaw' (language 'en')
>>    --  Playing 'digits/1.alaw' (language 'en')
>>
>> # The system claims to start MOH "default" which works elsewhere, but
>> the caller gets silence
>>    -- Started music on hold, class 'default', on DAHDI/2-1
>>  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
>> 'Parked/DAHDI/2-1'
>>
>
> Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but
> it has killed call quality on ISDN calls - I think it interferes with
> the software echo canceller somehow.
>
> Is there a ticket open on this? A patch to try?
>
> Thanks,
> Steve

For anyone searching/finding this issue, the patch here:

  https://issues.asterisk.org/view.php?id=17726

Applies to 1.6.2 with only a trivial tweak, and with minimal testing
is working here. We now get music on hold when a call is parked, even
when we are using res_timing_dahdi.so - Call quality remains high
under these circumstances too.

Thanks for the pointers.

Regards,
Steve

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[asterisk-users] Dahdi issue with Asterisk 1.8.0

[Timothy Legge]

Hi

I was using the delivered Ubuntu 1.6.x packages but I wanted to look at
gtalk integration so I downloaded, compiled and installed the source (after
removing the Ubuntu packages) have installed the following:

asterisk-1.8.0
dahdi-linux-complete-2.4.0+2.4.0
libpri-1.4.11.5

I copied my config back into place and most seems to work, but I cannot get
my phone that is plugged into the Wildcard TDM400P REV E/F card that I have
to work.

Basically, I don't hear the dial tone and Asterisk does not register off
hook events.  I have spent time reviewing my config but I don't see what the
issue is.

Is there anything I am missing, or can you suggest some additional things to
look at?

Tim

chan_dahdi.conf
grep -v "^;" /etc/asterisk/chan_dahdi.conf | grep -v "^$"

[trunkgroups]
[channels]
language=en
context=phones
signalling=fxo_ks
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
group=1
callgroup=1
pickupgroup=1

dahdi-channels.conf:

; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
;;; line="1 WCTDM/4/0 FXOKS"
signalling=fxo_ks
callerid="Channel 1" <4001>
mailbox=4001
group=5
context=phones
channel => 1
callerid=
mailbox=
group=
context=default

;;; line="2 WCTDM/4/1 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=incoming-local
channel => 2
callerid=
group=
context=default
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Re: [asterisk-users] Issue with MOH - Asterisk 1.4.17

[Napoleón Ernesto López Espinoza]

>
> It's would also be advised to use a much more recent version.  Asterisk
> 1.4.17 has many bugs and security issues that have been addressed in
> newer versions.  1.4.37 is the latest version from the 1.4 branch.  It's
> quite possible that whatever you're trying to fix is already fixed in
> that newer release.
>

I followed your advice and updated to 1.4.37. I'm still not able to get the
musiconhold module to work. From the asterisk CLI, when I type 'reload' I
get the following notices and warnings:

[Dec  7 08:11:08] NOTICE[24340]: cdr.c:1449 do_reload: CDR simple logging
enable
d.
[Dec  7 08:11:08] WARNING[24340]: res_smdi.c:1406 reload: No SMDI interfaces
wer
e specified to listen on, not starting SDMI listener.
[Dec  7 08:11:08] NOTICE[24340]: app_playback.c:458 reload: Reloading
say.conf
[Dec  7 08:11:13] NOTICE[24340]: indications.c:502
ast_unregister_indication_cou
ntry: Removed default indication country 'us'
[Dec  7 08:11:13] NOTICE[24340]: pbx_ael.c:4516 pbx_load_module: Starting
AEL
lo
ad process.
[Dec  7 08:11:13] NOTICE[24340]: pbx_ael.c:4523 pbx_load_module: AEL load
proces
s: calculated config file name '/etc/asterisk/extensions.ael'.
[Dec  7 08:11:13] NOTICE[24340]: pbx_ael.c:4531 pbx_load_module: AEL load
proces
s: parsed config file name '/etc/asterisk/extensions.ael'.
[Dec  7 08:11:13] NOTICE[24340]: pbx_ael.c:4534 pbx_load_module: AEL load
proces
s: checked config file name '/etc/asterisk/extensions.ael'.
[Dec  7 08:11:13] NOTICE[24340]: pbx_ael.c:4540 pbx_load_module: AEL load
proces
s: compiled config file name '/etc/asterisk/extensions.ael'.
[Dec  7 08:11:13] NOTICE[24340]: pbx_ael.c:4543 pbx_load_module: AEL load
proces
s: merged config file name '/etc/asterisk/extensions.ael'.
[Dec  7 08:11:13] NOTICE[24340]: pbx_ael.c:4546 pbx_load_module: AEL load
proces
s: verified config file name '/etc/asterisk/extensions.ael'.
[Dec  7 08:11:13] WARNING[24340]: chan_dahdi.c:12207 process_dahdi: Ignoring
any
changes to 'switchtype' (on reload)
[Dec  7 08:11:13] WARNING[24340]: chan_dahdi.c:12207 process_dahdi: Ignoring
any
changes to 'signalling' (on reload)
[Dec  7 08:11:13] WARNING[24340]: chan_dahdi.c:12207 process_dahdi: Ignoring
any
changes to 'rxwink' (on reload)
Any more ideas?

Regards,

Napoleon Lopez
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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Andrew Latham]

On Tue, Dec 7, 2010 at 1:34 PM, Mike  wrote:
> Thanks.  Too bad, FTP makes it much easier for the multi-tenant systems (IMO)
>
> Mike

https://wiki.asterisk.org/wiki/display/AST/Configuration+of+phoneprov.conf
will provide you with some information and very soon if not currently
the tlsbind will work well and allow a secure server for provisioning.

~~~ Andrew "lathama" Latham lath...@gmail.com ~~~

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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Danny Nicholas]

Using the python method, you could set up multiple provisions since the
"root" is the directory you start the command in
See this link
http://www.linuxjournal.com/content/tech-tip-really-simple-http-server-pytho
n
so if you had /tmp/set1, /tmp/set2 and /tmp/set3
you could do 
cd /tmp/set1
python -m SimpleHTTPServer 8081
cd /tmp/set2
python -m SimpleHTTPServer 8082
cd /tmp/set3
python -m SimpleHTTPServer 8083 

and have 3 provisions active.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 10:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

Hi Danny,

Maybe I should spend more time thinking about this.  FTP (on Polycom)
allowed me to easily "segment" my customers, give them different versions of
the firmware, etc.

Thanks everyone for your help.

Mike


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, December 07, 2010 11:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

Actually (this is also IMO), the HTTP makes life much easier.  You can use
the Asterisk http function to do this provisioning or even easier than that,
do the "python -m SimpleHTTPServer"

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 10:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

Thanks.  Too bad, FTP makes it much easier for the multi-tenant systems
(IMO)

Mike

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, December 07, 2010 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

On Tue, 2010-12-07 at 09:43 -0500, Mike wrote:
> Hi,
> 
>  
> 
> I`m not actually asking for a comparaison between the two, I have one 
> on hand and will make up my own mind.  But I can t find much info on 
> whether the Snom (370 to be exact) accepts FTP provisioning like the 
> Polycom (but few others) do.
> 
>  
> 
>  
> 
> Any Snom user can answer this one for me?
> 
>  
> 
> Mike
> 
AFAIK you can't use FTP for mass deployment purposes on Snom phones and is
backed up by this link

http://wiki.snom.com/Features/Mass_Deployment/Setting_Server

The HTTP provisioning does seem quite nifty though, I've looked into it but
not yet had the nod to actually develop a provisioning server but it does
seem pretty straight forward.


--
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Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Mike]

Hi Danny,

Maybe I should spend more time thinking about this.  FTP (on Polycom)
allowed me to easily "segment" my customers, give them different versions of
the firmware, etc.

Thanks everyone for your help.

Mike


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, December 07, 2010 11:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

Actually (this is also IMO), the HTTP makes life much easier.  You can use
the Asterisk http function to do this provisioning or even easier than that,
do the "python -m SimpleHTTPServer"

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 10:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

Thanks.  Too bad, FTP makes it much easier for the multi-tenant systems
(IMO)

Mike

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, December 07, 2010 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

On Tue, 2010-12-07 at 09:43 -0500, Mike wrote:
> Hi,
> 
>  
> 
> I`m not actually asking for a comparaison between the two, I have one 
> on hand and will make up my own mind.  But I can t find much info on 
> whether the Snom (370 to be exact) accepts FTP provisioning like the 
> Polycom (but few others) do.
> 
>  
> 
>  
> 
> Any Snom user can answer this one for me?
> 
>  
> 
> Mike
> 
AFAIK you can't use FTP for mass deployment purposes on Snom phones and is
backed up by this link

http://wiki.snom.com/Features/Mass_Deployment/Setting_Server

The HTTP provisioning does seem quite nifty though, I've looked into it but
not yet had the nod to actually develop a provisioning server but it does
seem pretty straight forward.


--
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Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Danny Nicholas]

Actually (this is also IMO), the HTTP makes life much easier.  You can use
the Asterisk http function to do this provisioning or even easier than that,
do the "python -m SimpleHTTPServer"

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 10:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

Thanks.  Too bad, FTP makes it much easier for the multi-tenant systems
(IMO)

Mike

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, December 07, 2010 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

On Tue, 2010-12-07 at 09:43 -0500, Mike wrote:
> Hi,
> 
>  
> 
> I`m not actually asking for a comparaison between the two, I have one 
> on hand and will make up my own mind.  But I can t find much info on 
> whether the Snom (370 to be exact) accepts FTP provisioning like the 
> Polycom (but few others) do.
> 
>  
> 
>  
> 
> Any Snom user can answer this one for me?
> 
>  
> 
> Mike
> 
AFAIK you can't use FTP for mass deployment purposes on Snom phones and is
backed up by this link

http://wiki.snom.com/Features/Mass_Deployment/Setting_Server

The HTTP provisioning does seem quite nifty though, I've looked into it but
not yet had the nod to actually develop a provisioning server but it does
seem pretty straight forward.


--
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Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Mike]

Thanks.  Too bad, FTP makes it much easier for the multi-tenant systems (IMO)

Mike

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, December 07, 2010 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

On Tue, 2010-12-07 at 09:43 -0500, Mike wrote:
> Hi,
> 
>  
> 
> I`m not actually asking for a comparaison between the two, I have one 
> on hand and will make up my own mind.  But I can t find much info on 
> whether the Snom (370 to be exact) accepts FTP provisioning like the 
> Polycom (but few others) do.
> 
>  
> 
>  
> 
> Any Snom user can answer this one for me?
> 
>  
> 
> Mike
> 
AFAIK you can't use FTP for mass deployment purposes on Snom phones and is 
backed up by this link

http://wiki.snom.com/Features/Mass_Deployment/Setting_Server

The HTTP provisioning does seem quite nifty though, I've looked into it but not 
yet had the nod to actually develop a provisioning server but it does seem 
pretty straight forward.


--
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Andrew Latham]

On Tue, Dec 7, 2010 at 11:43 AM, Mike  wrote:
> Hi,
>
>
>
> I`m not actually asking for a comparaison between the two, I have one on
> hand and will make up my own mind.  But I can’t find much info on whether
> the Snom (370 to be exact) accepts FTP provisioning like the Polycom (but
> few others) do.
>
>
>
>
>
> Any Snom user can answer this one for me?
>
>
>
> Mike

snom uses HTTP with optional HTTP Auth.
http://wiki.snom.com/Settings/user_auth_tag

You can also look at using SSl / TLS for configurations.  Polycom also
does this now as HTTP offers more options.


~~~ Andrew "lathama" Latham lath...@gmail.com ~~~

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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Ishfaq Malik]

On Tue, 2010-12-07 at 09:43 -0500, Mike wrote:
> Hi,
> 
>  
> 
> I`m not actually asking for a comparaison between the two, I have one
> on hand and will make up my own mind.  But I can’t find much info on
> whether the Snom (370 to be exact) accepts FTP provisioning like the
> Polycom (but few others) do.
> 
>  
> 
>  
> 
> Any Snom user can answer this one for me?
> 
>  
> 
> Mike
> 
AFAIK you can't use FTP for mass deployment purposes on Snom phones and
is backed up by this link

http://wiki.snom.com/Features/Mass_Deployment/Setting_Server

The HTTP provisioning does seem quite nifty though, I've looked into it
but not yet had the nod to actually develop a provisioning server but it
does seem pretty straight forward.


-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] [headset/mic] Volume too low + echo in *

[Gilles]

Hello,

I'm having the following problem when using a headset on XP
connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus
motherboard:

- Using any sound recorder (Windows', Audacity, XLite), the level is
just too low when speaking at a conversational level, even with the
microphone level pumped all the way up (line displayed totally flat in
Recorder)
http://img704.imageshack.us/img704/7981/headsetlowvolumeecho.jpg

- In addition, when making a call with XLite and Asterisk, I get a bit
of echo

- Same issues when trying with a different headset

- Enabling "Auto gain control AGC" in XLite makes no difference.

Any idea what can be done? Should I use a different soundcard?
Amplified headset? Can something be done in Asterisk about the echo?

Thank you.


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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[klitzing]

Hi!

> One thing to keep in mind: TFTP and FTP are very different things, 
> security-wise (none vs some)
> But I will definitely try to fudge my way with ftp://

Not sure if this helps, but the "rescue mode" for firmware updates does support 
FTP on the 
8xx phones:

http://wiki.snom.com/Firmware/Update/TFTP_Update

Philipp


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Re: [asterisk-users] 'Bookmarking' a place in a sound file

[Steve Edwards]

On Tue, 7 Dec 2010, David Cunningham wrote:

> Is it possible to somehow 'bookmark' a place in a sound file? That is, 
> the user presses a key while a sound file is playing and that point is 
> saved, and some time in the future we can play the same sound file and 
> tell it to start playing from that point.
> 
> This would be done within a perl AGI program.

The AGI command 'stream file' will return 'endpos' when interrupted with a 
keypress. You could then save that in a channel variable or a database.

A subsequent call to 'stream file' would include 'endpos' as the 'sample 
offset.'

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Mike]

Thanks Nicholas,

 

One thing to keep in mind: TFTP and FTP are very different things,
security-wise (none vs some)

 

But I will definitely try to fudge my way with ftp://

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, December 07, 2010 10:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 9:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

 

Thanks, that`s actually a useful document but it doesn't specify whether I
can do FTP provisioning.   I could assume I can`t, but I hoping it`s not the
answer.

 

Mike

 

Two things to consider

#1 to the best of my knowledge, 90+ percent of phones allow TFTP
provisioning

#2 if you can do an http://foo   string, you can probably do an
ftp://foo   one as well

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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 9:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

 

Thanks, that`s actually a useful document but it doesn't specify whether I
can do FTP provisioning.   I could assume I can`t, but I hoping it`s not the
answer.

 

Mike

 

Two things to consider

#1 to the best of my knowledge, 90+ percent of phones allow TFTP
provisioning

#2 if you can do an http://foo   string, you can probably do an
ftp://foo   one as well

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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Mike]

Thanks, that`s actually a useful document but it doesn't specify whether I
can do FTP provisioning.   I could assume I can`t, but I hoping it`s not the
answer.

 

Mike

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, December 07, 2010 9:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Snom (vs Polycom) - provisioning

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 8:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Snom (vs Polycom) - provisioning

 

Hi,

 

I`m not actually asking for a comparaison between the two, I have one on
hand and will make up my own mind.  But I can't find much info on whether
the Snom (370 to be exact) accepts FTP provisioning like the Polycom (but
few others) do.

 

 

Any Snom user can answer this one for me?

 

Mike

I'm a Polycom-mer but this link might give you the information to set up
your SNOM 370 using ftp.

http://www.3cx.com/sip-phones/Snom-provisioning.html

 

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Re: [asterisk-users] No MOH with parked call

[Steve Davies]

On 7 December 2010 14:17, Lee Archer  wrote:
> Hi, try unloading res_timing_dahdi.so then trying again.
>
> Lee
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
> Davies
> Sent: 07 December 2010 12:54
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] No MOH with parked call
>
> Hi,
>
> Has anybody else noticed that MOH does not play on parked calls in
> 1.6.2.14? Or is it just my setup? MOH seems to work in every other
> respect (Call Held or in-Queue), but when a call is parked, the logs
> show MOH being started, but the parked party hears nothing.
>
> The verbose logs show the following. Any thoughts on whet to check next?
>
> Thanks,
> Steve
>
>
> ### Call comes in here and is answered
>    -- SIP/snom360-0d6f answered DAHDI/2-1
>    -- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f",
> "0?done") in new stack
>    -- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f",
> "CHANNEL(musicclass)=m-default") in new stack
>    -- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f",
> "") in new stack
>
> ### Here the call is being blind transferred to the Park number
>    -- Started music on hold, class 'default', on DAHDI/2-1
>    -- Stopped music on hold on DAHDI/2-1
>  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
>    -- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack
>    -- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in
> new stack
>
> ### Not sure why I send "Ringing" here, but I tried NoOP() and
> Answer() too just in case
>    -- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack
>    -- Executing [...@local:4] Set("DAHDI/2-1",
> "CHANNEL(musicclass)=default") in new stack
>    -- Executing [...@local:5] Set("DAHDI/2-1",
> "CHANNEL(parkinglot)=default") in new stack
>    -- Executing [...@local:6] Goto("DAHDI/2-1",
> "parkedcalls_default,park,1") in new stack
>    -- Goto (parkedcalls_default,park,1)
>    -- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in
> new stack
>  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
> extension [parkedcalls_default] s, 1 in 90 seconds
>    -- Added extension '211' priority 1 to parkedcalls_default
> (0xbe2e528)
>
> # The "211" announcement is heard perfectly
>    --  Playing 'digits/2.alaw' (language 'en')
>  == Extension Changed 211[extensions] new state InUse for Notify User
> steve
>    --  Playing 'digits/1.alaw' (language 'en')
>    --  Playing 'digits/1.alaw' (language 'en')
>
> # The system claims to start MOH "default" which works elsewhere, but
> the caller gets silence
>    -- Started music on hold, class 'default', on DAHDI/2-1
>  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
> 'Parked/DAHDI/2-1'
>

Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but
it has killed call quality on ISDN calls - I think it interferes with
the software echo canceller somehow.

Is there a ticket open on this? A patch to try?

Thanks,
Steve

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Re: [asterisk-users] Snom (vs Polycom) - provisioning

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 07, 2010 8:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Snom (vs Polycom) - provisioning

 

Hi,

 

I`m not actually asking for a comparaison between the two, I have one on
hand and will make up my own mind.  But I can't find much info on whether
the Snom (370 to be exact) accepts FTP provisioning like the Polycom (but
few others) do.

 

 

Any Snom user can answer this one for me?

 

Mike

I'm a Polycom-mer but this link might give you the information to set up
your SNOM 370 using ftp.

http://www.3cx.com/sip-phones/Snom-provisioning.html

 

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[asterisk-users] Snom (vs Polycom) - provisioning

[Mike]

Hi,

 

I`m not actually asking for a comparaison between the two, I have one on
hand and will make up my own mind.  But I can't find much info on whether
the Snom (370 to be exact) accepts FTP provisioning like the Polycom (but
few others) do.

 

 

Any Snom user can answer this one for me?

 

Mike

 

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Re: [asterisk-users] No MOH with parked call

[Lee Archer]

Hi, try unloading res_timing_dahdi.so then trying again.

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Davies
Sent: 07 December 2010 12:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] No MOH with parked call

Hi,

Has anybody else noticed that MOH does not play on parked calls in
1.6.2.14? Or is it just my setup? MOH seems to work in every other
respect (Call Held or in-Queue), but when a call is parked, the logs
show MOH being started, but the parked party hears nothing.

The verbose logs show the following. Any thoughts on whet to check next?

Thanks,
Steve


### Call comes in here and is answered
-- SIP/snom360-0d6f answered DAHDI/2-1
-- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f",
"0?done") in new stack
-- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f",
"CHANNEL(musicclass)=m-default") in new stack
-- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f",
"") in new stack

### Here the call is being blind transferred to the Park number
-- Started music on hold, class 'default', on DAHDI/2-1
-- Stopped music on hold on DAHDI/2-1
  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
-- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack
-- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in
new stack

### Not sure why I send "Ringing" here, but I tried NoOP() and
Answer() too just in case
-- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack
-- Executing [...@local:4] Set("DAHDI/2-1",
"CHANNEL(musicclass)=default") in new stack
-- Executing [...@local:5] Set("DAHDI/2-1",
"CHANNEL(parkinglot)=default") in new stack
-- Executing [...@local:6] Goto("DAHDI/2-1",
"parkedcalls_default,park,1") in new stack
-- Goto (parkedcalls_default,park,1)
-- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in
new stack
  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
extension [parkedcalls_default] s, 1 in 90 seconds
-- Added extension '211' priority 1 to parkedcalls_default
(0xbe2e528)

# The "211" announcement is heard perfectly
--  Playing 'digits/2.alaw' (language 'en')
  == Extension Changed 211[extensions] new state InUse for Notify User
steve
--  Playing 'digits/1.alaw' (language 'en')
--  Playing 'digits/1.alaw' (language 'en')

# The system claims to start MOH "default" which works elsewhere, but
the caller gets silence
-- Started music on hold, class 'default', on DAHDI/2-1
  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
'Parked/DAHDI/2-1'

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Re: [asterisk-users] Version compatibility question...

[Kevin P. Fleming]

On 12/06/2010 08:12 PM, C F wrote:
> Thanks Kevin.
> Upto which version fo Dahdi works with 1.4.x?

If I understand your question properly, all versions of DAHDI are 
compatible with 1.4.x. All versions of DAHDI are backward compatible.

-- 
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Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] No MOH with parked call

[Steve Davies]

Hi,

Has anybody else noticed that MOH does not play on parked calls in
1.6.2.14? Or is it just my setup? MOH seems to work in every other
respect (Call Held or in-Queue), but when a call is parked, the logs
show MOH being started, but the parked party hears nothing.

The verbose logs show the following. Any thoughts on whet to check next?

Thanks,
Steve


### Call comes in here and is answered
-- SIP/snom360-0d6f answered DAHDI/2-1
-- Executing [...@macro-set-moh-call:1]
GotoIf("SIP/snom360-0d6f", "0?done") in new stack
-- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f",
"CHANNEL(musicclass)=m-default") in new stack
-- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f",
"") in new stack

### Here the call is being blind transferred to the Park number
-- Started music on hold, class 'default', on DAHDI/2-1
-- Stopped music on hold on DAHDI/2-1
  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
-- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack
-- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in
new stack

### Not sure why I send "Ringing" here, but I tried NoOP() and
Answer() too just in case
-- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack
-- Executing [...@local:4] Set("DAHDI/2-1",
"CHANNEL(musicclass)=default") in new stack
-- Executing [...@local:5] Set("DAHDI/2-1",
"CHANNEL(parkinglot)=default") in new stack
-- Executing [...@local:6] Goto("DAHDI/2-1",
"parkedcalls_default,park,1") in new stack
-- Goto (parkedcalls_default,park,1)
-- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in new stack
  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
extension [parkedcalls_default] s, 1 in 90 seconds
-- Added extension '211' priority 1 to parkedcalls_default (0xbe2e528)

# The "211" announcement is heard perfectly
--  Playing 'digits/2.alaw' (language 'en')
  == Extension Changed 211[extensions] new state InUse for Notify User steve
--  Playing 'digits/1.alaw' (language 'en')
--  Playing 'digits/1.alaw' (language 'en')

# The system claims to start MOH "default" which works elsewhere, but
the caller gets silence
-- Started music on hold, class 'default', on DAHDI/2-1
  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
'Parked/DAHDI/2-1'

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Re: [asterisk-users] Push central phone book to phones

[Steve Howes]

On 7 Dec 2010, at 11:35, Jonas Kellens wrote:
> When on a public server, I find this insecure.

Then secure it? Tie down by IP address, or some phones support the 
username:password@ in a URL.

S
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Re: [asterisk-users] Push central phone book to phones

[Jonas Kellens]

On 12/07/2010 12:18 PM, Andrew Thomas wrote:
> For the Yealink - you can use a 'remote' XML file.  The XML is stored on
> a web server and is retrieved by the phone every time you press the
> phones 'key'.  This has the advantage of not needing the directory to be
> pushed to the handset - and the handset always gets the latest version.
>
> Of course, the XML file needs to be kept up to date every time someone's
> name/extn changes.
>
> HTH

When on a public server, I find this insecure.

Jonas.

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Re: [asterisk-users] Push central phone book to phones

[Andrew Thomas]

For the Yealink - you can use a 'remote' XML file.  The XML is stored on
a web server and is retrieved by the phone every time you press the
phones 'key'.  This has the advantage of not needing the directory to be
pushed to the handset - and the handset always gets the latest version.

Of course, the XML file needs to be kept up to date every time someone's
name/extn changes.

HTH

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 03 December 2010 13:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Push central phone book to phones

On 12/02/2010 04:31 PM, Ishfaq Malik wrote:
> On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote:
>
>> On 12/02/2010 03:47 PM, Ishfaq Malik wrote:
>>  
>>> On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote:
>>>
>>>
 Hello,

 I have Snom, Cisco, Grandstream&   YeaLink phones.

 Is there a way to push a centralized phone book to these phones ??



 Kind regards,
 Jonas.
 -- 

  
>>> With Snom phones (and also Yealink I think) you can use centralised
LDAP
>>> directories on a server
>>>
>>>
>> This is a public server on the internet. I don't think I can use LDAP
to
>> push then ?
>>
>>
>> Kind regards,
>> Jonas.
>>  
> If you can set up and administer LDAP on the server you will be able
to
> use it on the Snom (and maybe Yealink) phones.
>

I can use different Organizational Units for different phone books ??


Kind regards,
Jonas.




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[asterisk-users] 'Bookmarking' a place in a sound file

[David Cunningham]

Hi all,

Is it possible to somehow 'bookmark' a place in a sound file? That is, the
user presses a key while a sound file is playing and that point is saved,
and some time in the future we can play the same sound file and tell it to
start playing from that point.

This would be done within a perl AGI program.

Thanks for any advice!

-- 
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http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
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[asterisk-users] DUNDi and Lua dialplan

[Guillaume Bour]

Hello,

I would like to known how to use DUNDi with a Lua dialplan ?

In extensions.conf, we should do like these:
|[lookupdundi]
switch => DUNDi/priv

[internal]
include => dundiextens
include => lookupdundi

exten => _,2,NoOp(calling ${EXTEN})
exten => _,n,Dial(SIP/${EXTEN})
exten => _,n,Hangup()|

priority 1 is either defined in dundiextens (local registered devices) 
or lookupdundi (remote)


But as in Lua there is no priority, we can't to this.
I found the following method working:

|extensions = {
internal = {
["_"] = function(c,e)
app.noop('lua:: dialing exten ' .. e)
-- Goto is not working, I need to use a Local 
channel

app.dial('Local/'..e..'@lookupdundi')
app.dial('SIP/'..e)
app.hangup()
end;
};
}|

But is this correct/the best one ?

Regards,
Guillaume

--
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10 bis, rue Lucien VOILIN - 92800 Puteaux

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Re: [asterisk-users] no audio on end-point when call is connected/bridged via PBX

[Robles Román , José Miguel]

> I am running Asterisk 1.8 on a cloud server.  I have had the
> same configuration running on a physical machine with a
> similar configuration.
> Thoughts?  I know I posted this yesterday but was hoping for
> some more creative comments!

If signalling works and audio don't, it probably has to do with phones behind 
NAT. It seems necessary to review the configuration of local routers.

Regards,
José Miguel

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