Re: [asterisk-users] Inquiry:How to convert *.wav files ?

[ABBAS SHAKEEL]

Hello Hadi
In beginning i also face this problem . I solved it by converting to SLN
format.

You also try to convert it to sln format.

this link might help you
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk





On Sat, Sep 26, 2009 at 10:44 AM, hadi motamedi wrote:

> Dear All
> Can you please do me favor and let me know how can I convert *.wav files
> into 32 bit 44 KHz ? Please be informed that I have specific sound files in
> *.wav format that I converted them into *.gsm format with the aid of the
> following command :
> #sox FR3.wav FR3.gsm
> It got through but the voice quality is poor . I need to convert the
> original *.wav sound files (their file attribute is reported as WAVE audio
> mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please
> help me .
> Thank you in advance
>
>
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-- 
Best Regards
Shakeel Abbas
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Re: [asterisk-users] SIP/WiFi handsets?

[Frank Bulk]

Note to those Americans scratching their heads over this: nano-BTS systems
are not so unusual in the Netherlands, unlike the USA.

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Thursday, September 24, 2009 7:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP/WiFi handsets?

On Thu, 2009-09-24 at 09:56 -0400, jon pounder wrote:
> Dean Collins wrote:
> 
> Earlier in the thread someone made a comment about using gsm since 
> everyone had gsm handsets already.
> 
> Can you explain in detail please ? (what hardware specifically, and how 
> does this actually work ?) My ignorant assumption is something like the 
> end user has a cell phone that actually works with 2 carriers - yours 
> and the "real" carrier.
> 

Your assumption is correct.
We set up our own wireless network on several locations / ships, using
nano-BTS systems. Sites were interconnected via VPN's and satelite
links.
And made a roaming agreement with other GSM-providers.
On location (Withing the reach of our own transmitters) you see our name
as gsm-provider and if you move away several kilometers, yoo switch
automagically to a national provider.

Word of caution, probably only viable for large organisations/companies

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Re: [asterisk-users] "multiple contexts for multiple locations"

[John A. Sullivan III]

On Fri, 2009-09-25 at 16:58 -0500, das sandesh wrote:
> Hi All,
> 
> I have a senario where we have multiple locations and all have the
> ability to call using 1NX pattern, so we have created multiple
> contexts so the outbound goes fine, but while transfer occurs (after
> picking the inbound call and transfer), it uses the first 1Nx
> priority patterned context, like if the 3rd location is making a
> transfer, but 1st location have the priority since it is declared
> first..so i am not able to adjust proper priorities based on the
> context..Is there a way to search based on the extension's
> context.since the extensions have the contexts based on the
> locations...

I'm not sure I fully understand the problem.  If it is similar to ours
we we needed to match outbound patterns and general sip patterns and
have multiple locations and contexts, we enforced the order of extension
processing by using include statements - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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[asterisk-users] OT - In which countries are ISDN subaddresses used ?

[Olivier]

Hi,

I've seen this ISDN subaddress feature added to libpri.
Which countries are using it ?
How is this billed ? Do you have to pay an extra to your telco to benefit
from this subaddresses ?

Cheers
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Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware

[Loic Didelot]

Thank you,
the mounting part was missing in my setup.

Loïc.



On Fri, 2009-09-25 at 20:01 +0300, Tzafrir Cohen wrote:
> On Fri, Sep 25, 2009 at 04:30:19PM +0200, Loic Didelot wrote:
> > # lsusb 
> > Bus 002 Device 003: ID e4e4:1160  
> > Bus 002 Device 001: ID :  
> > Bus 001 Device 003: ID 0403:e6c8 Future Technology Devices
> > International, Ltd 
> > Bus 001 Device 002: ID 0403:6001 Future Technology Devices
> > International, Ltd FT232 USB-Serial (UART) IC
> > Bus 001 Device 001: ID :  
> > # ls /proc/bus/usb -a
> > .  ..
> 
> Add the following line to /etc/fstab:
> 
> procbususb /proc/bus/usb usbfs defaults 0 0
> 
> 
> Then run:
> 
>   mount /proc/bus/usb
> 
> (or: 'mount -a')
> 
> Once that is in place, either disconnect and reconnect the Astribank, or
> use:
> 
>   /usr/share/dahdi/xpp_fxloader usb
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

 -- Goto (macro-user-callerid,s,18)
[Sep 25 13:32:44] -- Executing [...@macro-user-callerid:18] 
NoOp("DAHDI/1-1", "Using CallerID "xx" ") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:2] Set("DAHDI/1-1", 
"RingGroupMethod=none") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:3] Set("DAHDI/1-1", 
"VMBOX=100") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:4] Set("DAHDI/1-1", 
"EXTTOCALL=100") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:5] Set("DAHDI/1-1", 
"CFUEXT=") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:6] Set("DAHDI/1-1", 
"CFBEXT=") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:7] Set("DAHDI/1-1", 
"RT=25") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:8] 
Macro("DAHDI/1-1", "record-enable,100,IN") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-record-enable:1] 
GotoIf("DAHDI/1-1", "1?check") in new stack
[Sep 25 13:32:44] -- Goto (macro-record-enable,s,4)
[Sep 25 13:32:44] -- Executing [...@macro-record-enable:4] 
AGI("DAHDI/1-1", "recordingcheck,20090925-133244,1253899963.12") in new 
stack
[Sep 25 13:32:44] -- Launched AGI Script 
/var/lib/asterisk/agi-bin/recordingcheck
[Sep 25 13:32:44]  recordingcheck,20090925-133244,1253899963.12: Inbound 
recording not enabled
[Sep 25 13:32:44] -- AGI Script recordingcheck completed, 
returning 0
[Sep 25 13:32:44] -- Executing [...@macro-record-enable:5] 
MacroExit("DAHDI/1-1", "") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-exten-vm:9] 
Macro("DAHDI/1-1", "dial,25,t,100") in new stack
[Sep 25 13:32:44] -- Executing [...@macro-dial:1] GotoIf("DAHDI/1-1", 
"1?dial") in new stack
[Sep 25 13:32:44] -- Goto (macro-dial,s,3)
[Sep 25 13:32:44] -- Executing [...@macro-dial:3] AGI("DAHDI/1-1", 
"dialparties.agi") in new stack
[Sep 25 13:32:44] -- Launched AGI Script 
/var/lib/asterisk/agi-bin/dialparties.agi
[Sep 25 13:32:44]  dialparties.agi: Starting New Dialparties.agi
[Sep 25 13:32:44]  dialparties.agi: Caller ID name is 'xx' 
number is 'xx'
[Sep 25 13:32:44]  dialparties.agi: Methodology of ring is  'none'
[Sep 25 13:32:44] -- dialparties.agi: Added extension 100 to 
extension map
[Sep 25 13:32:44] -- dialparties.agi: Extension 100 cf is disabled
[Sep 25 13:32:44] -- dialparties.agi: Extension 100 do not disturb 
is disabled
[Sep 25 13:32:44]  dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
[Sep 25 13:32:44] -- dialparties.agi: dbset CALLTRACE/100 to xx
[Sep 25 13:32:44] -- dialparties.agi: Filtered ARG3: 100
[Sep 25 13:32:44] -- AGI Script dialparties.agi 
completed, returning 0
[Sep 25 13:32:44] -- Executing [...@macro-dial:7] Dial("DAHDI/1-1", 
"SIP/100,25,t") in new stack
[Sep 25 13:32:44]   == Using SIP RTP TOS bits 184
[Sep 25 13:32:44]   == Using SIP RTP CoS mark 5
[Sep 25 13:32:44] -- Called 100
[Sep 25 13:32:44] -- SIP/100-086aef08 is ringing
[Sep 25 13:33:09]   == Spawn extension (macro-dial, s, 7) exited 
non-zero on 'DAHDI/1-1' in macro 'dial'
[Sep 25 13:33:09]   == Spawn extension (macro-exten-vm, s, 9) exited 
non-zero on 'DAHDI/1-1' in macro 'exten-vm'
[Sep 25 13:33:09]   == Spawn extension (from-did-direct, 100, 1) exited 
non-zero on 'DAHDI/1-1'
[Sep 25 13:33:09] -- Hungup 'DAHDI/1-1'
chrislynn*CLI>

And this is if I hangup during voicemail:
[Sep 25 13:34:29] -- Starting simple switch on 'DAHDI/1-1'
[Sep 25 13:34:30] -- Executing [...@from-zaptel:1] NoOp("DAHDI/1-1", 
"Entering from-zaptel with DID == ") in new stack
[Sep 25 13:34:30] -- Executing [...@from-zaptel:2] 
Ringing("DAHDI/1-1", "") in new stack
[Sep 25 13:34:30] -- Executing [...@from-zaptel:3] Set("DAHDI/1-1", 
"DID=s") in new stack
[Sep 25 13:34:30] -- Executing [...@from-zaptel:4] NoOp("DAHDI/1-1", 
"DID is now s") in new stack
[Sep 25 13:34:30] -- Executing [...@from-zaptel:5] GotoIf("DAHDI/1-1", 
"1?zapok:notzap") in new stack
[Sep 25 13:34:30] -- Goto (from-zaptel,s,8)
[Sep 25 13:34:30] -- Executing [...@from-zaptel:8] NoOp("DAHDI/1-1", 
"Is a Zaptel Channel") in new stack
[Sep 25 13:34:30] -- Executing [...@from-zaptel:9] Set("DAHDI/1-1", 
"CHAN=1-1") in new stack
[Sep 25 13:34:30] -- Executing [...@from-zaptel:10] Set("DAHDI/1-1", 
"CHAN=1") in new stack
[Sep 25 13:34:30] -- Executin

Re: [asterisk-users] DAHDI disconnect supervision timing

[Tzafrir Cohen]

On Fri, Sep 25, 2009 at 11:19:39AM -0400, Stephen Brown Jr wrote:
> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
> FXO port and 1 FXS port. I have a POTS line from my phone company attached
> to the POTS line.
> 
> I have asked for "disconnect supervision" to be provisioned on my line and
> they claim to have added it. However, my scenario is as follows:
> 
> I receive a call, if the caller hangs up before hitting voice mail, the
> DAHDI channel is released as to be expected (evidenced from console
> messaging)
> If the call gets to voicemail and the caller hangs up during the greeting,
> no hangup condition is ever detected and I am greeted with a useless
> voicemail moments later.

I don't understand this.

Can you enable debug logging and provide log of a call that disconnect
successfully and a log of a call that fails to disconnect?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] How to remove peers from channels

[RSCL Mumbai]

On Fri, Sep 25, 2009 at 10:27 PM, Philipp Kempgen  wrote:

> RSCL Mumbai schrieb:
> > Pls see below output.
> > I would like to remove the last 3 peers.
> > How can I do this ?
>
> > [trixbox ~]# /usr/sbin/asterisk -rx "sip show channels"
>
> Use grep. (See `man grep`.)
>

I may not have explained my requirement well.

I do not wish to remove the peers from the listing.
I want the peers to not be there at all.

These peers (EyeBeam extensions) had connected to the Trixbox about 24+
hours ago.
At this moment, I do not have anyone connected to my Trixbox server from IP:
122.169. using extension 1006.
But I see 3 peers showing as connected from 122.169. using extension
1006.

Thx
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Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware

[Tzafrir Cohen]

On Fri, Sep 25, 2009 at 04:30:19PM +0200, Loic Didelot wrote:
> # lsusb 
> Bus 002 Device 003: ID e4e4:1160  
> Bus 002 Device 001: ID :  
> Bus 001 Device 003: ID 0403:e6c8 Future Technology Devices
> International, Ltd 
> Bus 001 Device 002: ID 0403:6001 Future Technology Devices
> International, Ltd FT232 USB-Serial (UART) IC
> Bus 001 Device 001: ID :  
> # ls /proc/bus/usb -a
> .  ..

Add the following line to /etc/fstab:

procbususb /proc/bus/usb usbfs defaults 0 0


Then run:

  mount /proc/bus/usb

(or: 'mount -a')

Once that is in place, either disconnect and reconnect the Astribank, or
use:

  /usr/share/dahdi/xpp_fxloader usb

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] How to remove peers from channels

[Philipp Kempgen]

RSCL Mumbai schrieb:
> Pls see below output.
> I would like to remove the last 3 peers.
> How can I do this ?

> [trixbox ~]# /usr/sbin/asterisk -rx "sip show channels"

Use grep. (See `man grep`.)


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] DAHDI disconnect supervision timing

[Danny Nicholas]

No joy, but a suggestion.  The default voicemail call is Voicemail(xxx,b) or
Voicemail(xxx,u) for busy or unavailable.  If you did Voicemail(xxx,s)
(silent) you could playback the voicemail greeting from the dialplan, then
check the line using the ChanisAvail function before launching
Voicemail(xxx,s).  Something like this;

-  exten => s,n,playback(greeting)

-  exten => s,n,Chanisavail(DAHDI/1)

-  exten => s,n,Gotoif($["${AVAILCHAN}" = "DAHDI/1"]?hungup)

-  exten => s,n,Voicemail(xxx,s)

-  exten => s,n(hungup),Hangup

 

YMMV, and not tested, but should work.

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Sent: Friday, September 25, 2009 10:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DAHDI disconnect supervision timing

 

Have you been able to find a satisfiable config? My biggest headache is the
useless voicemails being left if the caller hangs up during the greeting,
otherwise it appears to work as intended. 

Agree on the Zoloft, this is driving me nuts!

On 9/25/09 11:45 AM, Danny Nicholas wrote: 

If you're really going to pursue this, I'd buy stock in Zoloft - I've got a
TDM400 and TDP410 and they both drive me nuts on POTS issues.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Sent: Friday, September 25, 2009 10:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DAHDI disconnect supervision timing

 

I have, but I wanted to see if I could fix this problem before I started
experimenting with that. 

On 9/25/09 11:24 AM, Danny Nicholas wrote: 

Have you looked into minimum message length and/or silence parameters?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Jr
Sent: Friday, September 25, 2009 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DAHDI disconnect supervision timing

 

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line. 

I have asked for "disconnect supervision" to be provisioned on my line and
they claim to have added it. However, my scenario is as follows:

I receive a call, if the caller hangs up before hitting voice mail, the
DAHDI channel is released as to be expected (evidenced from console
messaging)
If the call gets to voicemail and the caller hangs up during the greeting,
no hangup condition is ever detected and I am greeted with a useless
voicemail moments later. 

I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
this same logic be applied to DAHDI somewhere? My theory is that the
"disconnect supervision" signal coming from the phone company may be less
than 1000ms. 

Desperately trying to fix this.

Thanks, 
Stephen

 
 
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Re: [asterisk-users] disable dtmf on SIP peer

[Giedrius Augys]

2009/9/25 Martin 

> rather you could
> disallow=alaw
> disallow=ulaw
> and set dmtfmode=inband
> since only g711 codec is clear enough to detect dtmf reliably
>
> Martin
>
>
> On Fri, Sep 25, 2009 at 10:30 AM, Giedrius Augys  wrote:
> > Hello,
> >
> >
> >I have one problem and I need to disable dtmf (disable rfc2833, info
> and
> > inband) on one (other peers must support dtmf) SIP peer . Is it possible?
> > Workaround would be use g729 codec with dtmfmode=inband.
> >
> > Maybe there is better solution?
> >
> > Thanks for help.
> >
> >
> > --
> > Pagarbiai  / Best Regards,
> > Giedrius Augys
> >
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> >
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Yes, I think it can work, but some agents are using SJ labs or other soft
phones, which doesn't support G729 codec.
I think it's not a best solution.
-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

Have you been able to find a satisfiable config? My biggest headache is 
the useless voicemails being left if the caller hangs up during the 
greeting, otherwise it appears to work as intended.


Agree on the Zoloft, this is driving me nuts!

On 9/25/09 11:45 AM, Danny Nicholas wrote:


If you're really going to pursue this, I'd buy stock in Zoloft -- I've 
got a TDM400 and TDP410 and they both drive me nuts on POTS issues...




*From:* asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Stephen Brown

*Sent:* Friday, September 25, 2009 10:41 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] DAHDI disconnect supervision timing

I have, but I wanted to see if I could fix this problem before I 
started experimenting with that.


On 9/25/09 11:24 AM, Danny Nicholas wrote:

Have you looked into minimum message length and/or silence parameters?



*From:* asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Stephen Brown Jr

*Sent:* Friday, September 25, 2009 10:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] DAHDI disconnect supervision timing

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card 
with 1 FXO port and 1 FXS port. I have a POTS line from my phone 
company attached to the POTS line.


I have asked for "disconnect supervision" to be provisioned on my line 
and they claim to have added it. However, my scenario is as follows:


I receive a call, if the caller hangs up before hitting voice mail, 
the DAHDI channel is released as to be expected (evidenced from 
console messaging)
If the call gets to voicemail and the caller hangs up during the 
greeting, no hangup condition is ever detected and I am greeted with a 
useless voicemail moments later.


I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, 
can this same logic be applied to DAHDI somewhere? My theory is that 
the "disconnect supervision" signal coming from the phone company may 
be less than 1000ms.


Desperately trying to fix this.

Thanks,
Stephen

  

  
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

I've tried the hanguponpolarityswitch parameter as well with no success :(

Any clues where in the DAHDI code I might find reference to disconnect 
supervision timing?

On 9/25/09 11:39 AM, Martin wrote:
> find the code in dahdi and put printk so you can see in dmesg or
> /var/log/messages
> if that gets ever detected
>
> also you may try hanguponpolarityswitch=yes in chan_dahdi.conf
>
> Martin
>
>
> On Fri, Sep 25, 2009 at 10:19 AM, Stephen Brown Jr
>   wrote:
>
>> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
>> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
>> FXO port and 1 FXS port. I have a POTS line from my phone company attached
>> to the POTS line.
>>
>> I have asked for "disconnect supervision" to be provisioned on my line and
>> they claim to have added it. However, my scenario is as follows:
>>
>> I receive a call, if the caller hangs up before hitting voice mail, the
>> DAHDI channel is released as to be expected (evidenced from console
>> messaging)
>> If the call gets to voicemail and the caller hangs up during the greeting,
>> no hangup condition is ever detected and I am greeted with a useless
>> voicemail moments later.
>>
>> I am using kewlstart signaling etc. I came across this page from Digium:
>>
>> http://kb.digium.com/entry/6/
>>
>> This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
>> this same logic be applied to DAHDI somewhere? My theory is that the
>> "disconnect supervision" signal coming from the phone company may be less
>> than 1000ms.
>>
>> Desperately trying to fix this.
>>
>> Thanks,
>> Stephen
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 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
>>
>>  
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Tilghman Lesher]

On Friday 25 September 2009 10:19:39 Stephen Brown Jr wrote:
> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
> FXO port and 1 FXS port. I have a POTS line from my phone company attached
> to the POTS line.
>
> I have asked for "disconnect supervision" to be provisioned on my line and
> they claim to have added it. However, my scenario is as follows:
>
> I receive a call, if the caller hangs up before hitting voice mail, the
> DAHDI channel is released as to be expected (evidenced from console
> messaging)
> If the call gets to voicemail and the caller hangs up during the greeting,
> no hangup condition is ever detected and I am greeted with a useless
> voicemail moments later.
>
> I am using kewlstart signaling etc. I came across this page from Digium:
>
> http://kb.digium.com/entry/6/
>
> This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
> this same logic be applied to DAHDI somewhere? My theory is that the
> "disconnect supervision" signal coming from the phone company may be less
> than 1000ms.

The equivalent header in DAHDI is include/dahdi/kernel.h, in the dahdi-linux
source package.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] disable dtmf on SIP peer

[Giedrius Augys]

2009/9/25 Martin 

> rather you could
> disallow=alaw
> disallow=ulaw
> and set dmtfmode=inband
> since only g711 codec is clear enough to detect dtmf reliably
>
> Martin
>
>
> On Fri, Sep 25, 2009 at 10:30 AM, Giedrius Augys  wrote:
> > Hello,
> >
> >
> >I have one problem and I need to disable dtmf (disable rfc2833, info
> and
> > inband) on one (other peers must support dtmf) SIP peer . Is it possible?
> > Workaround would be use g729 codec with dtmfmode=inband.
> >
> > Maybe there is better solution?
> >
> > Thanks for help.
> >
> >
> > --
> > Pagarbiai  / Best Regards,
> > Giedrius Augys
> >
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> >
>
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Hi,

  The problem is with Flash Hook.  On asterisk I've created call center, but
all agents are registered to other VoIP gw.
The problem appears, when one agent wants transfer call to other agent by
pressig Flash button. And then Asterisk and another VoIP starts playing MOH,
the client hears two different MOH on the same time. When second agent
answers the call, client hears only asterisk MOH, and agent silence.
Asterisk doesn't stop playing. So I want that asterisk ignores flash hook
and doesn't start playing MOH.
-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Danny Nicholas]

If you're really going to pursue this, I'd buy stock in Zoloft - I've got a
TDM400 and TDP410 and they both drive me nuts on POTS issues.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Sent: Friday, September 25, 2009 10:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DAHDI disconnect supervision timing

 

I have, but I wanted to see if I could fix this problem before I started
experimenting with that. 

On 9/25/09 11:24 AM, Danny Nicholas wrote: 

Have you looked into minimum message length and/or silence parameters?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Jr
Sent: Friday, September 25, 2009 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DAHDI disconnect supervision timing

 

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line. 

I have asked for "disconnect supervision" to be provisioned on my line and
they claim to have added it. However, my scenario is as follows:

I receive a call, if the caller hangs up before hitting voice mail, the
DAHDI channel is released as to be expected (evidenced from console
messaging)
If the call gets to voicemail and the caller hangs up during the greeting,
no hangup condition is ever detected and I am greeted with a useless
voicemail moments later. 

I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
this same logic be applied to DAHDI somewhere? My theory is that the
"disconnect supervision" signal coming from the phone company may be less
than 1000ms. 

Desperately trying to fix this.

Thanks, 
Stephen

 
 
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Re: [asterisk-users] disable dtmf on SIP peer

[Martin]

rather you could
disallow=alaw
disallow=ulaw
and set dmtfmode=inband
since only g711 codec is clear enough to detect dtmf reliably

Martin


On Fri, Sep 25, 2009 at 10:30 AM, Giedrius Augys  wrote:
> Hello,
>
>
>    I have one problem and I need to disable dtmf (disable rfc2833, info and
> inband) on one (other peers must support dtmf) SIP peer . Is it possible?
> Workaround would be use g729 codec with dtmfmode=inband.
>
> Maybe there is better solution?
>
> Thanks for help.
>
>
> --
> Pagarbiai  / Best Regards,
> Giedrius Augys
>
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

I have, but I wanted to see if I could fix this problem before I started 
experimenting with that.


On 9/25/09 11:24 AM, Danny Nicholas wrote:


Have you looked into minimum message length and/or silence parameters?



*From:* asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Stephen Brown Jr

*Sent:* Friday, September 25, 2009 10:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] DAHDI disconnect supervision timing

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card 
with 1 FXO port and 1 FXS port. I have a POTS line from my phone 
company attached to the POTS line.


I have asked for "disconnect supervision" to be provisioned on my line 
and they claim to have added it. However, my scenario is as follows:


I receive a call, if the caller hangs up before hitting voice mail, 
the DAHDI channel is released as to be expected (evidenced from 
console messaging)
If the call gets to voicemail and the caller hangs up during the 
greeting, no hangup condition is ever detected and I am greeted with a 
useless voicemail moments later.


I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, 
can this same logic be applied to DAHDI somewhere? My theory is that 
the "disconnect supervision" signal coming from the phone company may 
be less than 1000ms.


Desperately trying to fix this.

Thanks,
Stephen


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Re: [asterisk-users] disable dtmf on SIP peer

[Dave Fullerton]

Giedrius Augys wrote:
> Hello,
> 
> 
>I have one problem and I need to disable dtmf (disable rfc2833, info and
> inband) on one (other peers must support dtmf) SIP peer . Is it possible?
> Workaround would be use g729 codec with dtmfmode=inband.
> 
> Maybe there is better solution?
> 
> Thanks for help.
> 

Assuming you have control over the peer, simply set the peer to use 
rfc2833 and have asterisk listen for info (or other way around) on that 
peer.

-Dave

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Re: [asterisk-users] DAHDI disconnect supervision timing

[Martin]

find the code in dahdi and put printk so you can see in dmesg or
/var/log/messages
if that gets ever detected

also you may try hanguponpolarityswitch=yes in chan_dahdi.conf

Martin


On Fri, Sep 25, 2009 at 10:19 AM, Stephen Brown Jr
 wrote:
> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
> FXO port and 1 FXS port. I have a POTS line from my phone company attached
> to the POTS line.
>
> I have asked for "disconnect supervision" to be provisioned on my line and
> they claim to have added it. However, my scenario is as follows:
>
> I receive a call, if the caller hangs up before hitting voice mail, the
> DAHDI channel is released as to be expected (evidenced from console
> messaging)
> If the call gets to voicemail and the caller hangs up during the greeting,
> no hangup condition is ever detected and I am greeted with a useless
> voicemail moments later.
>
> I am using kewlstart signaling etc. I came across this page from Digium:
>
> http://kb.digium.com/entry/6/
>
> This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
> this same logic be applied to DAHDI somewhere? My theory is that the
> "disconnect supervision" signal coming from the phone company may be less
> than 1000ms.
>
> Desperately trying to fix this.
>
> Thanks,
> Stephen
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> Register Now: http://www.astricon.net
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Re: [asterisk-users] disable dtmf on SIP peer

[Jeff LaCoursiere]

On Fri, 25 Sep 2009, Giedrius Augys wrote:

> Hello,
>
>
>   I have one problem and I need to disable dtmf (disable rfc2833, info and
> inband) on one (other peers must support dtmf) SIP peer . Is it possible?
> Workaround would be use g729 codec with dtmfmode=inband.

I don't think that would *disable* it, I think it would simply make it 
unpredictable :)

>
> Maybe there is better solution?
>

What are you really trying to accomplish?  Keep the user from being able 
to navigate menus?  There should be diaplan/context ways to keep them from 
doing internal things, but if you are trying to keep them from interacting 
with external services I am not sure.

j

> Thanks for help.
>
>
> -- 
> Pagarbiai  / Best Regards,
> Giedrius Augys
>

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[asterisk-users] disable dtmf on SIP peer

[Giedrius Augys]

Hello,


   I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.

Maybe there is better solution?

Thanks for help.


-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Danny Nicholas]

Have you looked into minimum message length and/or silence parameters?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Jr
Sent: Friday, September 25, 2009 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DAHDI disconnect supervision timing

 

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line. 

I have asked for "disconnect supervision" to be provisioned on my line and
they claim to have added it. However, my scenario is as follows:

I receive a call, if the caller hangs up before hitting voice mail, the
DAHDI channel is released as to be expected (evidenced from console
messaging)
If the call gets to voicemail and the caller hangs up during the greeting,
no hangup condition is ever detected and I am greeted with a useless
voicemail moments later. 

I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
this same logic be applied to DAHDI somewhere? My theory is that the
"disconnect supervision" signal coming from the phone company may be less
than 1000ms. 

Desperately trying to fix this.

Thanks, 
Stephen

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[asterisk-users] How to remove peers from channels

[RSCL Mumbai]

Pls see below output.
I would like to remove the last 3 peers.
How can I do this ?

Thx
Vai

[trixbox ~]# /usr/sbin/asterisk -rx "sip show channels"
Peer User/ANRCall ID  Seq (Tx/Rx)  Format
Hold Last Message
192.168.1.126(None)  MjkzYjNiMmY  00101/4  0x0 (nothing)
No   Rx: REGISTER
64.154.41.1067552235573  147111b67e3  11342/0  0x0 (nothing)No
81.201.84.45 3866719789  3TUNX3-CPYZ  00101/00102  0x100 (g729)
No   Rx: ACK
195.189.173.10   301241893b37329b407  19037/0  0x0 (nothing)No
81.201.84.45 4172802551  NC75LK-XAU5  00101/00102  0x100 (g729)
No   Rx: ACK
192.168.1.13 10100cb8ef0d570  00102/0  0x280100 (g729|
No   Tx: ACK
192.168.1.13 10053cc07973759  00102/0  0x280100 (g729|
No   Tx: ACK
81.201.84.45 7709498956  ECSTS5-MU5R  00101/00102  0x100 (g729)
No   Rx: ACK
81.201.84.45 9147616530  VTTE3C-CN2Z  00101/00102  0x100 (g729)
No   Rx: ACK
81.201.84.45 9414471279  GC2W4P-ZPWN  00101/00102  0x100 (g729)
No   Rx: ACK
192.168.1.13 1007080f5e47519  00102/0  0x280100 (g729|
No   Tx: ACK
81.201.84.45 9858786358  CQQ5M7-ZM4F  00101/00102  0x100 (g729)
No   Rx: ACK
81.201.84.45 3189496064  FDK6CY-2LSF  00101/00102  0x100 (g729)
No   Rx: ACK
192.168.1.13 10160758ea9a349  00102/0  0x0 (nothing)No
(d)  Tx: ACK
122.169.113.145  1006379b29497d0  00102/0  0x0 (nothing)
No   Init: NOTIFY
122.169.113.145  10063a4fc558695  00102/0  0x0 (nothing)
No   Init: NOTIFY
122.169.113.145  10067678828b011  00102/0  0x0 (nothing)
No   Init: NOTIFY
17 active SIP channels
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[asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown Jr]

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line.

I have asked for "disconnect supervision" to be provisioned on my line and
they claim to have added it. However, my scenario is as follows:

I receive a call, if the caller hangs up before hitting voice mail, the
DAHDI channel is released as to be expected (evidenced from console
messaging)
If the call gets to voicemail and the caller hangs up during the greeting,
no hangup condition is ever detected and I am greeted with a useless
voicemail moments later.

I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
this same logic be applied to DAHDI somewhere? My theory is that the
"disconnect supervision" signal coming from the phone company may be less
than 1000ms.

Desperately trying to fix this.

Thanks,
Stephen
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Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware

[Loic Didelot]

# lsusb 
Bus 002 Device 003: ID e4e4:1160  
Bus 002 Device 001: ID :  
Bus 001 Device 003: ID 0403:e6c8 Future Technology Devices
International, Ltd 
Bus 001 Device 002: ID 0403:6001 Future Technology Devices
International, Ltd FT232 USB-Serial (UART) IC
Bus 001 Device 001: ID :  
# ls /proc/bus/usb -a
.  ..


Loic


On Thu, 2009-09-24 at 13:21 +0300, Tzafrir Cohen wrote:
> On Thu, Sep 24, 2009 at 10:54:17AM +0200, Loic Didelot wrote:
> > Not sure,
> > how can I check, but older astribanks work pretty fine on that system.
> 
> ls /proc/bus/usb
> 
> What is the output of:
> 
>   lsusb
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] Asterisk Manager Problem

[Andrei Verovski (aka MacGuru)]

A small clarification - a package I'm referring to is called Asterisk GUI, not 
Asterisk Manager. Sorry for mistype.

On Friday 25 September 2009 01:00:53 pm andreil1 wrote:
> Hi!
>
> I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
> repository), configured http.conf and manager.conf according to the
> manual.
>
> However, whenever I try to connect to Asterisk manager via web browser
> (http://192.168.0.1: , where xxx port defined in asterisk ->
> http.conf), I've got this error:
>
> Not Implemented
> Attempt to use unimplemented / unsupported method
> Asterisk Server
>
> Downgrading Asterisk to version 1.6, or even installing on another
> SuSE box did not help.
>
> Anyone have any idea what is wrong?
>
> Many thanks in advance for any suggestion(s)

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Re: [asterisk-users] Choppy sound, SIP calls within LAN

[John A. Sullivan III]

On Fri, 2009-09-25 at 13:01 +0300, andreil1 wrote:
> Hi!
> 
> I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE  
> repository). As a clients I use XLite on Mac, all on the same LAN.  
> Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM  
> and plenty of disk space on LEVEL 5 RAID.
> 
> Calls to another SIP server (also asterisk) hosted by another company  
> are 100% OK, so it is clearly problem with my server setup.
> 
> Background music (before pickup) runs fine, but transmitted voice  
> sound is very choppy, no matter of which codec I use.
> 
> I have searched over net, and implemented one by one every reasonable  
> receipt found, including.
> 
> highpriority = yes
> internal_timing = yes
> 
> transmit_silence = no
> 
> nat = yes
> localnet=192.168.0.0/255.255.0.0
> externip = xx.xx.xx.xx
> 
> dtmfmode=rfc2833
> 
> Downgrading asterisk did not solved problem, too.
> 
> Anyone please help if possible..
> 
> Many thanks in advance for any suggestion(s).
> 

My first guess would be a network problem.  Is there something different
in the network path between the users and the hosted Asterisk server
versus the users and the internal Asterisk server? Have you implement
some form of CoS / QoS internally (one should)? If you run a continuous
ping from a user to the internal Asterisk server, is there any packet
loss or congestion (indicated by widely varying response times)? Just a
few thoughts - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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[asterisk-users] Choppy sound, SIP calls within LAN

[andreil1]

Hi!

I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE  
repository). As a clients I use XLite on Mac, all on the same LAN.  
Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM  
and plenty of disk space on LEVEL 5 RAID.

Calls to another SIP server (also asterisk) hosted by another company  
are 100% OK, so it is clearly problem with my server setup.

Background music (before pickup) runs fine, but transmitted voice  
sound is very choppy, no matter of which codec I use.

I have searched over net, and implemented one by one every reasonable  
receipt found, including.

highpriority = yes
internal_timing = yes

transmit_silence = no

nat = yes
localnet=192.168.0.0/255.255.0.0
externip = xx.xx.xx.xx

dtmfmode=rfc2833

Downgrading asterisk did not solved problem, too.

Anyone please help if possible..

Many thanks in advance for any suggestion(s).

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[asterisk-users] Asterisk Manager Problem

[andreil1]

Hi!

I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE  
repository), configured http.conf and manager.conf according to the  
manual.

However, whenever I try to connect to Asterisk manager via web browser  
(http://192.168.0.1: , where xxx port defined in asterisk ->  
http.conf), I've got this error:

Not Implemented
Attempt to use unimplemented / unsupported method
Asterisk Server

Downgrading Asterisk to version 1.6, or even installing on another  
SuSE box did not help.

Anyone have any idea what is wrong?

Many thanks in advance for any suggestion(s)

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Re: [asterisk-users] Error When Using Postgresql Schema WithRealtime Sip

[stephen.hindmarch]

OK, have done. Issue ID 0015963.

Steve Hindmarch
BT Design 


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tilghman Lesher
> Sent: 24 September 2009 15:11
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Error When Using Postgresql Schema
> WithRealtime Sip
> 
> On Thursday 24 September 2009 05:06:02 stephen.hindma...@bt.com wrote:
> > I have investigated further and found that it is a bug in ODBC, not
> > Asterisk. The SQLColumns function, which asterisk uses to describe
> the
> > table, does not return any columns when the table name includes the
> > schema specification. You can show this by using isql to do "help
> table"
> > which returns info about all the columns, and then "help
> public.table"
> > which returns nothing. As chan_sip seems to be the only application
> that
> > tests the structure of the table before writing to it this is why
> > REGISTER fails.
> >
> > When I have time I will chase up ODBC and see if the issue is
tracked
> > there. Do you still want me to raise it as an issue on bugtracker?
> 
> Yes, I want you to raise this on the bugtracker, and no, this is not a
> bug
> in ODBC, but a deficiency in my code.  Since you tracked this down to
> the
> code in res_odbc.c, I might as well tell you that the first two NULL
> sets of
> arguments (NULL, 0) are for specifying the catalog and schema,
> respectively,
> of the database table, and it is because I never bothered parsing the
> schema
> out of the tablename that this does not work.
> 
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
> 
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Re: [asterisk-users] Polycom push application for microbrowser

[randulo]

Hi,

Sorry about posted a protected link, I forgot we'd closed the site to
spammers since we don't use it anymore. The useful content was
re-posted in our list.

---
URI dialing on Polycom phones

http://tr.im/polyapp

One of the applications originally posted in the VoIP Users Conference
site by Dave Van Ginneken
---


I encourage anyone interested in this kind of thing to join us in this
quiet and friendly mailing list (which is a Google Group so I found
the link to Dave's application).  It's another way we, the VoIP users
community, keep in touch and share.


Also of possible interest:

June 2009: Polycom Applications with Mike Seto, Polycoms VP of market
and business development for Polycom’s voice communications division.

 http://VUC.me/2009/02/polycom-applications-with-mike-seto/

Hope some of this helps. Dave's scripts work and they point up the
serious barriers to writing apps for Polycom: the docs are horrible to
wade through and Polycom does not make an effort to make application
programming easy of efficient. AT worst, someone should write a book
with example. The microbrowser can be a useful addition to the phone.
I use it to display an RSS feed. That script should also be in the VUC
list somewhere.

Regards,

Randy

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[asterisk-users] ignore flash hook

[Giedrius Augys]

Hello,

   I've created Call Center with Asterisk (1.6.0.5). Call Center's agents
are not Asterisk SIP user's, but other's voip gw SIP 5 class users.
Everything works fine, except when one agent wants transfer call to other
agent. They do it with flash hook. So and two voip gws (Asterisk and other
gw) detects this flash and both starts playing MOH. The transfer is
unsuccessfully, cause client doesn't hear anything when seconds answers the
call.

  My question is, is it possible (and how to do it), that asterisk ignores
flash hook in queue? Maybe is it possible to ignore this flash at all?

Thanks for help

-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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