Re: [asterisk-users] music on hold

[Tzafrir Cohen]

On Tue, Nov 11, 2008 at 05:53:39PM +1100, Lee, John (Sydney) wrote:
 The reason is your audio file is too high quality.
 
 Asterisk can only play back audio file of 4000Hz.

8000Hz (and also: mono, 16 bit sample rate).

What is the output of:  file path/to/sound.wav

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] TE410P alarms stay RED with 1.4.22

[Louis-David Mitterrand]

Hi,

I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but
then my TE410P alarms stay RED and no zap channels can be created, even
if they are correctly listed by zap show channels. I tried adding
dahdichanname = no to asterisk.conf's [options] to no effect.

Going back to 1.4.21.2 brings my alarms back to OK.

This is with zaptel 1.4.12.1.

-- 
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[asterisk-users] view the current calls and their codec

[nik600]

Hi to all.

Is possible with the Asterisk 1.4 cli view the  current calls and their codec?

Thanks to all
-- 
/*/
nik600
http://www.kumbe.it

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Re: [asterisk-users] view the current calls and their codec

[Tim Nelson]

[EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'

assuming you want SIP... substitute sip for iax2 if you prefer...

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- nik600 [EMAIL PROTECTED] wrote:

 Hi to all.
 
 Is possible with the Asterisk 1.4 cli view the  current calls and
 their codec?
 
 Thanks to all
 -- 
 /*/
 nik600
 http://www.kumbe.it
 
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[asterisk-users] OT: Polycom Firmware available (by accident?)

[Dave Fullerton]

Not sure if Polycom is changing their policy or if this is an accident, 
but you can actually download SIP 3.1.1 right from their web site.

Anyone looking for firmware should get it now before it disappears.

SIP app and release notes can be found here:
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html

-Dave

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Re: [asterisk-users] DNS A queries for channel

[samuel]

First of all apologies for not catching up that conversation, I hadn't found
it before. Thanks for the pointer.

I wouldn't recommend using wildcards for this problem because sometimes the
DNS query contains domain names which are valid and might be used for other
actions (such as registering,mail,...).
But if it's the only solution because no one knows where the DNS query is
performed and the subdomains are not used for other things...do it at your
own risk!!!

Thanks for all hints!!!
Samuel

2008/11/11 Philipp Kempgen [EMAIL PROTECTED]

 Matt Riddell schrieb:

  All solved by a caching dns server - achieved in debian by:
 
apt-get install bind9
 
  followed by change nameserver to 127.0.0.1 in resolv.conf

 http://lists.digium.com/pipermail/asterisk-users/2008-September/218764.html


   Philipp Kempgen

 --
 http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
 Amooma GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Dave Fullerton]

Jared Smith wrote:
 On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
 Anyone looking for firmware should get it now before it disappears.
 
 It's my understanding that this isn't a fluke, but that Polycom has
 indeed changed their policy and will no longer you to go through your
 reseller to get the latest and greatest firmware.
 

That's awesome!

I had wondered that but since I hadn't seen links for 3.1.0B or the new 
BootROM's it made me a little suspicious.

Thanks for the clarification.

-Dave

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Re: [asterisk-users] view the current calls and their codec

[nik600]

And if i have an h323 configuration?

Thanks


On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED] wrote:
 [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'

 assuming you want SIP... substitute sip for iax2 if you prefer...

 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105

 - nik600 [EMAIL PROTECTED] wrote:

 Hi to all.

 Is possible with the Asterisk 1.4 cli view the  current calls and
 their codec?

 Thanks to all
 --
 /*/
 nik600
 http://www.kumbe.it

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-- 
/*/
nik600
http://www.kumbe.it

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Re: [asterisk-users] GEN-GEN and Manual Ring-Down (MRD)?

[Jorge Mendoza]

Raj Jain wrote:
 On Mon, Nov 10, 2008 at 6:56 AM, Tony Mountifield
 [EMAIL PROTECTED] wrote:
   
 Does anyone here know anything about GEN-GEN analogue circuits, also
 known as Manual Ring-Down (MRD)? Apparently they are widely used in
 Hoot'n'Holler systems for financial dealer-boards.

 I have been asked to try and interface to such circuits, and have been
 having great difficulty locating any specifications for the interface.

 Apparently, they are always-on 2-wire analogue circuits with no tip
 voltage or loop current, and on-demand superimposed ringing voltage in
 either direction for signalling (to do nothing more than get the remote
 end's attention).

 I was wondering whether it is possible to adapt an FXS or FXO port to
 operate in such a mode, but I'm not optimistic.
 

 Your understanding of MRD is correct (these are nailed-up connections
 with only ring-gen capability). I've personally not tried this w/
 Asterisk FXS/FXO ports but If you can make it work that way pls. let
 us know.

 --
 Raj Jain

   
The MRD telephones uses local battery, that is the reason why they do
not have loop current (central battery). Any adaptor to a FXS  circuit
is useless because there are not any signalling to indicate on/off hook.
Just the initial manual ringing.
Then working on FXS ports is almost impossible or very expensive.
Another approach is to use E/M signalling. The audio channel could be
open permanently and transmit the ringing over the E/M wires.  You need
a  ringing detector and a ringing generator in both sides. Take care of
isolate the audio channels from the ringing current. I do that many
years ago on PCM muxes with E/M interfaces.
The Multitech gateways have E/M interfaces, but never tested under this
conditions.
Obviously, the easy way is to use two standard sets working as hotline.

Jorge Mendoza

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Re: [asterisk-users] view the current calls and their codec

[Tim Nelson]

'oh323 show channels' I would assume... I don't have a box handy with h323 
loaded to verify.

Check http://astrecipes.net/index.php?n=89

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- nik600 [EMAIL PROTECTED] wrote:

 And if i have an h323 configuration?
 
 Thanks
 
 
 On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED]
 wrote:
  [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'
 
  assuming you want SIP... substitute sip for iax2 if you prefer...
 
  Tim Nelson
  Systems/Network Support
  Rockbochs Inc.
  (218)727-4332 x105
 
  - nik600 [EMAIL PROTECTED] wrote:
 
  Hi to all.
 
  Is possible with the Asterisk 1.4 cli view the  current calls and
  their codec?
 
  Thanks to all
  --
  /*/
  nik600
  http://www.kumbe.it
 
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 -- 
 /*/
 nik600
 http://www.kumbe.it
 
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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Dave Fullerton]

Tilghman Lesher wrote:
 On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
 Jared Smith wrote:
 On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
 Anyone looking for firmware should get it now before it disappears.
 It's my understanding that this isn't a fluke, but that Polycom has
 indeed changed their policy and will no longer you to go through your
 reseller to get the latest and greatest firmware.
 That's awesome!

 I had wondered that but since I hadn't seen links for 3.1.0B or the new
 BootROM's it made me a little suspicious.
 
 I suspect that they do not want to support beta releases, so they don't
 make those generally available.  Also, the page you posted is for a fairly
 new phone, which I suspect already has the newest BootROM, so that download
 would be completely irrelevant.  If you instead looked at the page for a
 Polycom Soundpoint 430, for example, you would have found the BootROM, as
 this model was released with an earlier version.
 
 http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip430.html
 

I used the 450 link because it was the first one I came to. I had looked 
at the other phones pages just in case and they didn't have them either. 
It only offers 4.0.0 for download and not 4.1.0 or 4.1.2

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Re: [asterisk-users] TE410P alarms stay RED with 1.4.22

[Laurent Caron]

Bonjour Louis-David,

Asterisk envoie-t-il le signal au boitier pour le failover ?

Laurent



Le 11 nov. 08 à 08:49, Louis-David Mitterrand [EMAIL PROTECTED] 
g a écrit :

 Hi,

 I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22  
 but
 then my TE410P alarms stay RED and no zap channels can be created,  
 even
 if they are correctly listed by zap show channels. I tried adding
 dahdichanname = no to asterisk.conf's [options] to no effect.

 Going back to 1.4.21.2 brings my alarms back to OK.

 This is with zaptel 1.4.12.1.

 -- 
 http://www.critikart.net

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Re: [asterisk-users] view the current calls and their codec

[nik600]

thanks a lot!

On Tue, Nov 11, 2008 at 6:06 PM, Tim Nelson [EMAIL PROTECTED] wrote:
 'oh323 show channels' I would assume... I don't have a box handy with h323 
 loaded to verify.

 Check http://astrecipes.net/index.php?n=89

 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105

 - nik600 [EMAIL PROTECTED] wrote:

 And if i have an h323 configuration?

 Thanks


 On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED]
 wrote:
  [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'
 
  assuming you want SIP... substitute sip for iax2 if you prefer...
 
  Tim Nelson
  Systems/Network Support
  Rockbochs Inc.
  (218)727-4332 x105
 
  - nik600 [EMAIL PROTECTED] wrote:
 
  Hi to all.
 
  Is possible with the Asterisk 1.4 cli view the  current calls and
  their codec?
 
  Thanks to all
  --
  /*/
  nik600
  http://www.kumbe.it
 
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 --
 /*/
 nik600
 http://www.kumbe.it

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-- 
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nik600
http://www.kumbe.it

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Re: [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz.

[Remi Quezada]

I remember having that issue once and I fixed it by changing a
configuration in the motherboard BIOS.  It was related to the SATA hard
drive mode it was running on.  The default configuration was set to
legacy mode I believe and when I changed it to enhanced mode the lost of
interrupts problem I was having went away.  I had a Supermicro
motherboard in case you were curious. 

Remi Quezada

Giorgio Incantalupo wrote:
 Hi,

 I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy 
 is working fine but for some reason I cannot.
 The two machines have the same kernel, motherboard, the same gcc version 
 and the same zaptel 1.4.8. On the second machine zaptel compiles without 
 errors and ztdummy.ko is generated but when I modprobe it I get the 
 following error in messages:
 rtc: lost some interrupts at 1024Hz.
 Since modules sizes are slighly different I copied the working ztdummy 
 on the new machine so the files are the same.nothing changes!!

 Any help is appreciated.

 Thank you.

   


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Re: [asterisk-users] TE410P alarms stay RED with 1.4.22

[Laurent Caron]

Hi,

I'm sorry i didn't check the recipient while replying.

Sorry about the noise...

Laurent



Le 11 nov. 08 à 11:44, Laurent Caron [EMAIL PROTECTED] a écrit :

 Bonjour Louis-David,

 Asterisk envoie-t-il le signal au boitier pour le failover ?

 Laurent









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Re: [asterisk-users] changing the size of voice packets

[Pezhman Lali]

is any command , shows the current rate of each channel?
 

--- On Mon, 11/10/08, Kristian Kielhofner [EMAIL PROTECTED] wrote:
From: Kristian Kielhofner [EMAIL PROTECTED]
Subject: Re: [asterisk-users] changing the size of voice packets
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Monday, November 10, 2008, 11:10 PM

On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov
[EMAIL PROTECTED] wrote:

 If the packetisation durations are different between endpoints, the SDP
 offer/answer should fail with a 488 Not Acceptable Here. 
Right?


Depends?

What is the status of maxptime in Asterisk?

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Doug]

At 11:58 11/11/2008, Dave Fullerton wrote:
 Tilghman Lesher wrote:
  On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
  Jared Smith wrote:
  On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
  Anyone looking for firmware should get it now before it disappears.
  It's my understanding that this isn't a fluke, but that Polycom has
  indeed changed their policy and will no longer you to go through your
  reseller to get the latest and greatest firmware.
  That's awesome!

Older stuff:
http://tinyurl.com/OlderPolycomFirmware
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/previous_voip_software.html
 


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[asterisk-users] Dial outside number using the E1 Link

[fateme fatah]

Hi: 
I've configured an asterisk server with A102d sangoma's card and the E1 link.I 
want to dial outside number using the E1 Link.How can I dial a phone number?Is 
this true? 
exten = 123,1,Dial(ZAP/1/phone number) 
I'd appreciate any help.


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Re: [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz

[JR Richardson]

 I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy
 is working fine but for some reason I cannot.
 The two machines have the same kernel, motherboard, the same gcc version
 and the same zaptel 1.4.8. On the second machine zaptel compiles without
 errors and ztdummy.ko is generated but when I modprobe it I get the
 following error in messages:
 rtc: lost some interrupts at 1024Hz.
 Since modules sizes are slighly different I copied the working ztdummy
 on the new machine so the files are the same.nothing changes!!

Your mother boards are probably not 100% the same, maybe a chipset is
newer and causing an interrupt problem.  I've seen this before.  Put
'acpi=off' in your kernel boot parameter line in the grub menu.lst
like this:

title   Debian GNU/Linux, kernel 2.6.18-686
root(hd0,0)
kernel  /boot/vmlinuz-2.6.18--686 root=/dev/sda1 ro acpi=off
initrd  /boot/initrd.img-2.6.18-686
savedefault

Reboot, and that should do it.

JR
-- 
JR Richardson
Engineering for the Masses

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Re: [asterisk-users] Forcing repacketization on SIP to SIP call

[Richard Brady]

JT

Thanks for this detailed response. It's clear I have some more homework to
do before going anywhere near Mantis, but I will follow up either way.

Regards,
Richard

On Tue, Oct 28, 2008 at 9:02 PM, John Todd [EMAIL PROTECTED] wrote:


 This seems like a transcoding issue, and the RTP code may not be
 clever enough to understand that a repacketization is transcoding
 and therefore lets the media flow directly and/or passes the RTP
 packets through without examining or modifying them.  This could be an
 error in the way RTP transcoding is handled - put on your superhero
 bugtracking cape and post to Mantis!

 I'd suggest that you document this clearly, and put it on the
 bugs.digium.com system if you've tried all possible iterations of
 allow= and deny= for getting this media to transcode.   It would seem
 that alaw:20 is different than alaw:40, and if you've found that
 they are treated as equal then there seems to be a problem.  While not
 explicitly stated in the doc/rtp-packetization.txt file, it does
 seem that several things are true:

  - it seems that if a remote sender is sending 40ms packets, and you
 have not explicitly denied 40ms packets, that Asterisk should accept
 those packets.  This seems to work.

  - if you explicitly have deny=all and then allow=alaw:20 in a
 peer definition, it should be the case that Asterisk takes whatever
 audio stream and transcode it for the remote peer in that format (and
 in the SDP Asterisk should offer a ptime and maxptime based on the
 default and highest ptime acceptable, in this case 20 for both.)
 Is this broken?

  - if a remote host sends you a ptime that is not defined or
 defaulted in the list of allow= codec choices for that peer (or
 globally) then the call should be refused just like it would be with
 any other codec mismatch.  (Of course, if you don't have a deny=all
 as the first statement in your peer codec list, Asterisk should let
 anything through since that's the way those ACLs work.  I mention this
 only as a caution for reporting problems that might not be problems.)
 Is this broken?


 This problem is actually fairly important when we start talking about
 scale.  All RTP-based systems start to experience bottlenecks
 introduced by Packets-Per-Second limits on hardware interfaces.  The
 upper limit of performance starts to be more bound to throughput on
 interfaces and kernel drivers, rather than in the higher-layer code.
 PPS, not megabits per second, becomes the number to beat.  If you can
 get RTP packets to go from 20ms to 40ms, it doubles the size of the
 packet and effectively halves the number of packets you're sending on
 your interface, which _could_ lead to doubling of total numbers of
 calls as the limits of interface buffering are reached in the near
 future.   Even if you're just doing this on one leg of a looped call,
 this still could reduce your overall PPS by 25%, which is nothing to
 sniff at.  Of course, I'm assuming that the load introduced by re-
 packetizing different packet delays is not significant - this could be
 a false assumption.

 JT


 ---
 John Todd
 [EMAIL PROTECTED]+1-256-428-6083
 Asterisk Open Source Community Director

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Re: [asterisk-users] view the current calls and their codec

[Anthony Francis]

core show channels shows all channels and the first part of the ouput 
gives you the technology:

*CLI core show channels
Channel  Location State   
Application(Data)
SIP/xxx   (None)   Up  Bridged 
Call(Zap/2-1)
Zap/2-1  [EMAIL PROTECTED] Up  Dial(SIP/xxx

to get more output add the keyword verbose, to make it machine 
parse-able add the keyword concise.

Tim Nelson wrote:
 'oh323 show channels' I would assume... I don't have a box handy with h323 
 loaded to verify.

 Check http://astrecipes.net/index.php?n=89

 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105

 - nik600 [EMAIL PROTECTED] wrote:

   
 And if i have an h323 configuration?

 Thanks


 On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED]
 wrote:
 
 [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'

 assuming you want SIP... substitute sip for iax2 if you prefer...

 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105

 - nik600 [EMAIL PROTECTED] wrote:

   
 Hi to all.

 Is possible with the Asterisk 1.4 cli view the  current calls and
 their codec?

 Thanks to all
 --
 /*/
 nik600
 http://www.kumbe.it

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-- 
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP


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Re: [asterisk-users] Voicemail IMAP ./configure error

[c james]

Mark Michelson wrote:
 c james wrote:
 Mark Michelson wrote:
 c james wrote:
 I have c-client installed on a 64bit system running Gentoo.  I am trying
 to run configure so I can test the IMAP voicemail functionality. But

 asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap

 just gives me the following error.

 checking for gnutls_bye in -lgnutls... no
 checking for UW IMAP Toolkit c-client library... no
 checking for system c-client library.. no
 configure: ***
 configure: *** The UW IMAP Toolkit installation on this system appears
 to be broken.
 configure: *** Either correct the installation, or run configure
 configure: *** including --without-imap.

 c-client is installed.

 voicemail1 asterisk-1.4.22 # equery files c-client
 [ Searching for packages matching c-client... ]
 * Contents of net-libs/c-client-2006k:
 /usr/include/imap/c-client.h
   ... bunch of others
 /usr/include/imap/utf8aux.h
 /usr/lib64/c-client.a
 /usr/lib64/libc-client.a - c-client.a
 /usr/lib64/libc-client.so.1.0.0

 Interesting output there.

 If you specify --with-imap=/usr/src/imap then that means that the source 
 for 
 the imap toolkit is located at /usr/src/imap. It appears though, that only 
 the 
 c-client source is located there (or perhaps just the headers), and that 
 causes 
 the configure script to fail. If you specify just --with-imap with no 
 argument 
 or --with-imap=system then the configure script will try to find the 
 c-client 
 library and include files in common places where distributions tend to 
 install them.

 I'm guessing, though, that you did not download and compile the imap 
 toolkit 
 yourself and that you had Gentoo do it for you. The installation directory 
 for 
 the headers is different than where most distros place them. Most put the 
 c-client header files in /usr/include/c-client instead of /usr/include/imap.

 My suggestions for possible fixes are

 1) Try reconfiguring with just --with-imap or with --with-imap=system 
 instead of 
 specifying a directory. I'm suspecting this will not work properly because 
 of 
 the directory where the header files are, though.

 2) If step 1 fails like I think it will, then try moving the .h files from 
 /usr/include/imap to /usr/include/c-client and rerun the configure script 
 --with-imap and see if that helps. I suspect this will work. If it does, I 
 can 
 modify the configure script so that we search in the imap/ directory as 
 well as 
 the c-client directory for header files.

 If things still fail after those two steps, then respond with the section 
 from 
 the config.log file which displays the failure that occurred when searching 
 for 
 imap support.

 Mark Michelson


 You are correct, c-client was installed through the Gentoo portage
 command of

 emerge c-client

 Neither of the two suggestions worked.  Here is the relevant output from
 config.log

 configure:18552: checking for UW IMAP Toolkit c-client library
 configure:18630: gcc -o conftest -g -O2
 -I/usr/src/asterisk-1.4.22/../imap-2004g/c-client  conftest.c
 /usr/src/asterisk-1.4.22/../imap-2004g/c-client/c-client.a  5
 gcc: /usr/src/asterisk-1.4.22/../imap-2004g/c-client/c-client.a: No such
 file or directory
 conftest.c:145:22: error: c-client.h: No such file or directory

 
 Yuck. That check for the imap-2004g directory bugs me. It's not anything 
 you've 
 done, but a seemingly arbitrary decision that was made when the original IMAP 
 support was merged. The thing is, if a working IMAP installation is not found 
 in 
 that imap-2004g directory, the configure script is supposed to be smart 
 enough 
 to try to switch to the system-installed c-client library instead. Was there 
 any 
 further output down below what you have shown me that mentions something like 
 Checking for system c-client library...? If so, could you post the 
 config.log 
 output from that section?
 
 Mark Michelson
 

I attached the entire config.log


config.log.tgz
Description: application/compressed-tar
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[asterisk-users] music on hold

[Uros Djokic]

Hi,

You can convert your music files in 8000 hz and mono with sox command like
this
sox yourfile.wav -r 8000 -c 1 yournewfile,wav resample -ql

Uros

-- 
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To redistribute modified program under condition that new user has all 4
freedoms.
Richard M. Stallman
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Re: [asterisk-users] Server for 25-30 phones, sip trunks over the net

[Ben Hauger]

Well, we're running an asterisk 1.4.x system with a te220 span adapter 
(T1 PRI). 83 internal SIP peers, mostly Polycom Soundpoint IP series 
phones. It's a single-CPU dual-core Pentium E6420. The OS is CentOS 4.5 
(x86_64). Seems to work well, though it's only busy with call switching, 
voicemail, and call recording.

Cheers,
Ben

nb wrote:
 What cpu/memmory configurations have people had good luck with for a
 small office asterisk server... (polycom's/poe linksys)

 Dell's got their $200 SC440 going again until the 12th... and I'm
 thinking it might be just the ticket...

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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Vinícius Fontes]

Downloading right now, thank you very much for sharing it with us.


Atenciosamente,

Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
 
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000

- Dave Fullerton [EMAIL PROTECTED] escreveu:

 Not sure if Polycom is changing their policy or if this is an
 accident, 
 but you can actually download SIP 3.1.1 right from their web site.
 
 Anyone looking for firmware should get it now before it disappears.
 
 SIP app and release notes can be found here:
 http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html
 
 -Dave
 
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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Doug Lytle]

Dave Fullerton wrote:
 Not sure if Polycom is changing their policy or if this is an accident, 
   

As far as I've seen it reported here, it's a policy change.

Doug


-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?

[Tzafrir Cohen]

On Mon, Nov 10, 2008 at 07:34:34PM -0500, Jim Duda wrote:
 I've been having trouble with making outbound calls to my 
 TELCO from a TDM400 card (FXS KS signalling) after upgrading
 from 1.6-beta9 to 1.6.0.  The problem is completely intermittent.
 
 When it fails, I get this message:
 [Nov  9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 
 'DAHDI' (cause 0 - Unknown)

Can you enable debug logging? Do you see any message about the casue for
that?

 
 At some point, it starts working, but I don't know what is 
 triggering asterisk to start working.  But I have a theory below.
 
 I have instrumented the code with debugging and narrowed it 
 down to some code in chan_dahdi where it appears to be checking
 for hookstate.  I can actually resolve this issue by changing
 a return code in the available function in chan_dahdi.c
 
 This led me to look at dahdi show channel 4
 
 It appears that when outbound calls are working, 
 the Hookstate displays Offhook.
 
 It appears that when outobund calls are not working,
 the Hookstate displays Onhook.
 
 Can anyone tell me what the normal state of an FXS line attached
 to a standard TELCO should be when no call is in progress and
 when a call is in progress?
 
 Can anyone tell me what causes an FXS line attached to a
 standard TELCO to transition to Off Hook state?  It seems
 to me that the state would transition between off hook and
 on hook as a call is in progress or idle respectively.
 
 Thanks,
 
 Jim
 
 asterisk*CLI dahdi show channel 4
 Channel: 4LI 
 File Descriptor: 21
 Span: 1
 Extension: 
 Dialing: no
 Context: incoming
 Caller ID: 
 Calling TON: 0
 Caller ID name: 
 Mailbox: 100
 Destroy: 0
 InAlarm: 0
 Signalling Type: FXS Kewlstart
 Radio: 0
 Owner: None
 Real: None
 Callwait: None
 Threeway: None
 Confno: -1
 Propagated Conference: -1
 Real in conference: 0
 DSP: no
 Busy Detection: no
 TDD: no
 Relax DTMF: no
 Dialing/CallwaitCAS: 0/0
 Default law: ulaw
 Fax Handled: no
 Pulse phone: no
 DND: no
 Echo Cancellation:
128 taps
(unless TDM bridged) currently OFF
 Actual Confinfo: Num/0, Mode/0x
 Actual Confmute: No
 Hookstate (FXS only): Offhook -- Working State
 
 
 My chan_dahdi.conf:
 
 ;[pstn]
 mailbox=100
 mwimonitor=fsk
 mwilevel=512
 mwimonitornotify=/usr/local/sbin/zapnotify.sh
 faxdetect=incoming
 signalling=fxs_ks
 context=incoming
 callwaiting=yes
 callwaitingcallerid=yes
 echocancel=yes
 echotraining=no
 echocancelwhenbridged=no
 channel = 4
 
 ;[fax]
 ; FAX machine connected here
 faxdetect=no
 signalling=fxo_ks
 context=internal
 channel = 1

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Asterisk CDR Error ??

[Ruddy Gbaguidi]

hi all
do you guys know why asterisk sometimes, in the cdr put the dst (the 
extension) number in the src ??
I have 4 digit extensions (DID) and sometimes, the same values if found 
in the src that usually have the calling user caller id.
Thanks


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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Tilghman Lesher]

On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
 Jared Smith wrote:
  On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
  Anyone looking for firmware should get it now before it disappears.
 
  It's my understanding that this isn't a fluke, but that Polycom has
  indeed changed their policy and will no longer you to go through your
  reseller to get the latest and greatest firmware.

 That's awesome!

 I had wondered that but since I hadn't seen links for 3.1.0B or the new
 BootROM's it made me a little suspicious.

I suspect that they do not want to support beta releases, so they don't
make those generally available.  Also, the page you posted is for a fairly
new phone, which I suspect already has the newest BootROM, so that download
would be completely irrelevant.  If you instead looked at the page for a
Polycom Soundpoint 430, for example, you would have found the BootROM, as
this model was released with an earlier version.

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip430.html

-- 
Tilghman

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[asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

[Karl Fife]

In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'.  
Use the NEW ulaw/alaw codec's (slower, but cleaner)

By slower does this mean more 'expensive', or does it instead mean that there 
will be more algorithmic latency?  Both?  Can anyone speak to the relative 
increases?

With regard to accuracy, can anyone speak to what kind of situation might 
demonstrate the benefit of the new algorithm?  i.e. transcoding, SpanDSP, 
Analog interfaces (DAHDI) etc.

Thanks
Karl
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Re: [asterisk-users] music on hold

[邱磊]

^_^
asterisk should Encoding voice in 8KHZ ,16k bits,mono8
,i have formate the .wav files.
thank you for your advice,best regard.

2008-11-11 



邱磊 



发件人： Peter Evans 
发送时间： 2008-11-11  14:57:36 
收件人： Asterisk Users Mailing List - Non-Commercial Discussion 
抄送： 
主题： Re: [asterisk-users] music on hold 
 
On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote:
 hii guys:
   i get the message from the asterisk:
Started music on hold, class 'default', on Local/[EMAIL 
 PROTECTED],1
 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: 
 Unexpected freqency 11025
 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open 
 format wav
 [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 
 ast_moh_files_next: Unable to open file 
 '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory
 -- Stopped music on hold on Local/[EMAIL PROTECTED],1

TOM SKYPE + 163.com = I guess your first language is mandarin.

You know, life would be so much easier if people would read error messages and
think for a moment. Undoubtably, there are times when the error message is 
cryptic
beyond mortal understanding and you need to summon an elder thing to help you
read it, but this one is not of that ilk.
'/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory

This could be a tough one. 

If you can't solve this without help, should you really be playing with ancient
scrolls of wisdom in the first place?


P

ps:  没有??的文件或目?

but that probably doesnt work because I am using a Japanese font ^^;


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Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

[Steve Totaro]

On Tue, Nov 11, 2008 at 4:19 PM, Karl Fife [EMAIL PROTECTED] wrote:

  In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'.
 Use the NEW ulaw/alaw codec's (slower, but cleaner)

 By slower does this mean more 'expensive', or does it instead mean that
 there will be more algorithmic latency?  Both?  Can anyone speak to the
 relative increases?

 With regard to accuracy, can anyone speak to what kind of situation might
 demonstrate the benefit of the new algorithm?  i.e. transcoding, SpanDSP,
 Analog interfaces (DAHDI) etc.

 Thanks
 Karl



Interesting

Alaw and Ulaw algorithms slower and cleaner?  Slower would imply problems
with FAX or whatever.

Not sure how uncompressed audio could be cleaner and why on earth one would
want it slower?

What's next?  The new, improved, under new management, SLIN?

PS, I am sure (75% anyways) that this new algorithm is good for
something.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] DNS A queries for channel

[Matt Riddell]

On 11/11/2008 10:48 p.m., samuel wrote:
 So far I've updated a few machines (1.4.22) and the DNS queries are reduced
 to a minimum, at least haven't seen DNS channel queries...

Slightly more useful to me, does anyone know the patch that fixed it, so 
I can update machines I don't want to upgrade to 1.4.22?

Olle? Maybe?  :)

-- 
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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Re: [asterisk-users] DNS A queries for channel

[samuel]

2008/11/10 Matt Riddell [EMAIL PROTECTED]

 On 11/11/2008 1:34 a.m., samuel wrote:
  Folks,
 
  I'm tracing this error and looks like newer 1.4 (1.4.22) does not behave
 as
  I stated (no DNS for channel domainname) and it must have been solved in
 the
  way from my versions to the newer.
 
  I'll update to the newer versions and confirm it.

 Fantastic if that's the case.  I've seen quite a few people complaining
 about this and actually had an upstream DNS provider change IP :)

 All solved by a caching dns server - achieved in debian by:

  apt-get install bind9

 followed by change nameserver to 127.0.0.1 in resolv.conf


But this does not solve the problem, it's just a global work around.

Asterisk still makes the queries and if there's some problem in the DNS it
will continue blocking the first time the authoritative DNS crashes. The
problem is that since the channel is a random string, the queries can NOT
be cached (everytime the domain name is different) and the DNS timeout still
occur for every DNS channel query and the asterisk will block at some call
rate.

So far I've updated a few machines (1.4.22) and the DNS queries are reduced
to a minimum, at least haven't seen DNS channel queries...

Anyone familiar with this knows whether this issue is solved in 1.6?

Best regards,
Samuel.
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Re: [asterisk-users] DNS A queries for channel

[Tilghman Lesher]

On Tuesday 11 November 2008 05:17:30 Matt Riddell wrote:
 On 11/11/2008 10:48 p.m., samuel wrote:
  So far I've updated a few machines (1.4.22) and the DNS queries are
  reduced to a minimum, at least haven't seen DNS channel queries...

 Slightly more useful to me, does anyone know the patch that fixed it, so
 I can update machines I don't want to upgrade to 1.4.22?

http://svn.digium.com/view/asterisk?view=revrevision=133649

-- 
Tilghman

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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Jared Smith]

On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
 Anyone looking for firmware should get it now before it disappears.

It's my understanding that this isn't a fluke, but that Polycom has
indeed changed their policy and will no longer you to go through your
reseller to get the latest and greatest firmware.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] music on hold

[邱磊]

thank you，Ihave get the solution。
asterisk should Encoding voice in 8KHZ ,16k bits,mono8 ^^


2008-11-11 



邱磊 



发件人： Lee, John (Sydney) 
发送时间： 2008-11-11  14:55:00 
收件人： Asterisk Users Mailing List - Non-Commercial Discussion 
抄送： 
主题： Re: [asterisk-users] music on hold 
 
The reason is your audio file is too high quality.
Asterisk can only play back audio file of 4000Hz.
 



From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ??
Sent: Tuesday, 11 November 2008 5:35 PM
To: asterisk-users
Subject: [asterisk-users] music on hold
 
hii guys:
  i get the message from the asterisk:
   Started music on hold, class 'default', on Local/[EMAIL 
PROTECTED],1
[2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected 
freqency 11025
[2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open 
format wav
[2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: 
Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such 
file or directory
-- Stopped music on hold on Local/[EMAIL PROTECTED],1
 
 
  how can i solve the issue? thanks
 
2008-11-11 



$Bn9b}(J 
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Re: [asterisk-users] music on hold

[Jeff LaCoursiere]

On Tue, 11 Nov 2008, Peter Evans wrote:

 On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote:
  hii guys:
i get the message from the asterisk:
 Started music on hold, class 'default', on Local/[EMAIL 
  PROTECTED],1
  [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: 
  Unexpected freqency 11025
  [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open 
  format wav
  [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 
  ast_moh_files_next: Unable to open file 
  '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or 
  directory
  -- Stopped music on hold on Local/[EMAIL PROTECTED],1

   TOM SKYPE + 163.com = I guess your first language is mandarin.

   You know, life would be so much easier if people would read error 
 messages and
   think for a moment. Undoubtably, there are times when the error message 
 is cryptic
   beyond mortal understanding and you need to summon an elder thing to 
 help you
   read it, but this one is not of that ilk.

   '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or 
 directory

   This could be a tough one.

   If you can't solve this without help, should you really be playing with 
 ancient
   scrolls of wisdom in the first place?


Isn't it horribly embarrasing when you publicly trash someone and are
WRONG?  Why do people feel the need to be so cruel in the first place?

j

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[asterisk-users] help with call with no sound via PSTN

[César García]

Hello guys, I am having some problems with calls comming from the PSTN
lines, when somebody calls people can't hear me, but I can hear them,  every
day I have to do a /etc/init.d/asterisk stop  /etc/init.d/dahdi restart to
have calls with sound again, wich cli dubug commands can I use to see what
is going on, here I have my chan_dahdi.conf and sip.conf, I am using 1.6

Thanks a lot!

chan_dahdi.conf

trunkgroups]
[channels]
Group=1
context=incoming
signalling=fxs_ks
rxwink=300
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
jbenable=no
jbmaxsize=200
jbresyncthreshold=1000
useincomingcalleridondahditransfer=yes
;callerid=asrecived
rxgain=0.0
txgain=0.0
immediate=no
busydetect=yes
busycount=5
hidecallerid=no
callgroup=1
pickupgroup=1
channel = 1-24

sip.conf

[general]
disallow=all
allow=gsm
allow=ulaw
language=es

[sets](!)
type=friend
secret=1000
host=dynamic
;call-limit=1

[1109](sets)
[EMAIL PROTECTED]
context=oficina

[1110](sets)
[EMAIL PROTECTED]
context=oficina

[](sets)
[EMAIL PROTECTED]
context=oficina-jefatura

[1112](sets)
[EMAIL PROTECTED]
context=oficina

[1113](sets)
[EMAIL PROTECTED]
context=oficina

[1114](sets)
[EMAIL PROTECTED]
context=oficinaVoIP

[1115](sets)
[EMAIL PROTECTED]
context=oficina

[1116](sets)
[EMAIL PROTECTED]
context=oficina
[1117](sets)
[EMAIL PROTECTED]
context=oficinaVoIP

-- 
http://celord.blogspot.com/
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Re: [asterisk-users] Inbound/Outbound undesired behavior

[Lenz Emilitri]

My suggestion is to have an agent log-off from the ACD system (or at least
pause) before attempting outbound. You can achieve that with some handy
macro that might do this transparently before the call is placed and when it
terminates.

Just my two cents,

l.



2008/11/5 Ricardo Melendez [EMAIL PROTECTED]

 Hi to all, I need some help, I have an Asterisk Server in a small call
 center, for inbound calls I setup a Queue in queues.conf and their
 respective Agents in agents.conf, but when an Agent is calling out and a
 call is coming from PSTN the call is send to that agents which have a call
 in progress.

 How I can fix this in order to have only one call at a time.

 I think in limitonpeer and call-limit but the documentation says that the
 sip user can have 2 calls with this parameters (1 for inbound and 1 for
 Outbound and this is the behavior I don't want).


 Thanks in advance.

 Ricardo MR






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-- 
Loway - home of QueueMetrics - http://queuemetrics.com
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[asterisk-users] play file from url

[Mike Clark]

I would like to do something like:

exten = s,1,playback(http://my.server.com/file.wav)

I tested and it does not work. It seems highly likely that someone would 
already have done this one way or another. I know I could do a system 
wget and then play the local file, but wanted something a bit more elegant.

Thanks,

Mike Clark

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Re: [asterisk-users] DNS A queries for channel

[Philipp Kempgen]

Matt Riddell schrieb:

 All solved by a caching dns server - achieved in debian by:
 
   apt-get install bind9
 
 followed by change nameserver to 127.0.0.1 in resolv.conf

http://lists.digium.com/pipermail/asterisk-users/2008-September/218764.html


   Philipp Kempgen

-- 
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Amooma GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] play file from url

[Singer X.J. Wang]

Mike Clark wrote:

I would like to do something like:

exten = s,1,playback(http://my.server.com/file.wav)

I tested and it does not work. It seems highly likely that someone would 
already have done this one way or another. I know I could do a system 
wget and then play the local file, but wanted something a bit more elegant.


Thanks,

Mike Clark

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One issue with that is latency...

--
*Singer Wang*
/System and Database Engineer/
The Pythian Group

Office: (613) 565-8696 x298
Toll Free:  (877) 798-4426 x298
Fax:(613) 565-8710
Email:  [EMAIL PROTECTED]
MSN:[EMAIL PROTECTED]
Yahoo:  pythianwang
AIM:pythianwang
ICQ:201253
Gadu-Gadu:  6817795
Tencent QQ: 858310404

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fn:Singer Wang
n:Wang;Singer
email;internet:[EMAIL PROTECTED]
tel;work:(613) 565-8696 x298
x-mozilla-html:TRUE
version:2.1
end:vcard

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Re: [asterisk-users] TE410P alarms stay RED with 1.4.22

[Louis-David Mitterrand]

On Tue, Nov 11, 2008 at 09:49:14AM +0100, Louis-David Mitterrand wrote:
 Hi,
 
 I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but
 then my TE410P alarms stay RED and no zap channels can be created, even
 if they are correctly listed by zap show channels. I tried adding
 dahdichanname = no to asterisk.conf's [options] to no effect.
 
 Going back to 1.4.21.2 brings my alarms back to OK.

OK false alarm here: we use an isdnguard device that needs an additional
res_watchdog.c file (bristuff patch). Once added it works.

Let's hope 1.4.22 will solve our random crashes and system resources
hogging...

-- 
http://www.critikart.net

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[asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz.

[Giorgio Incantalupo]

Hi,

I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy 
is working fine but for some reason I cannot.
The two machines have the same kernel, motherboard, the same gcc version 
and the same zaptel 1.4.8. On the second machine zaptel compiles without 
errors and ztdummy.ko is generated but when I modprobe it I get the 
following error in messages:
rtc: lost some interrupts at 1024Hz.
Since modules sizes are slighly different I copied the working ztdummy 
on the new machine so the files are the same.nothing changes!!

Any help is appreciated.

Thank you.

-- 

_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FGA srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172  


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Re: [asterisk-users] Dial outside number using the E1 Link

[Pezhman Lali]

Dear Fateme
two good refrences:

http://articles.techrepublic.com.com/2415-1035_11-94140.html

and

http://www.trixbox.org/forums/vendor-forums-certified/sangoma/solved-sangoma-101d-card-trixbox-asterisk-1-4-19-1

hope to help u
best
Pezhman

--- On Tue, 11/11/08, fateme fatah [EMAIL PROTECTED] wrote:
From: fateme fatah [EMAIL PROTECTED]
Subject: [asterisk-users] Dial outside number using the E1 Link
To: asterisk-users@lists.digium.com
Date: Tuesday, November 11, 2008, 3:29 PM

Hi: 
I've configured an asterisk server with A102d sangoma's card and the E1 link.I 
want to dial outside number using the E1 Link.How can I dial a phone number?Is 
this true? 
exten = 123,1,Dial(ZAP/1/phone number) 
I'd appreciate any help.


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Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

[Steve Murphy]

On Tue, 2008-11-11 at 15:19 -0600, Karl Fife wrote:
 In Asterisk 1.6, there is an option to use the 'new g.711
 algorithm'.  
 Use the NEW ulaw/alaw codec's (slower, but cleaner)
  
 By slower does this mean more 'expensive', or does it instead mean
 that there will be more algorithmic latency?  Both?  Can anyone speak
 to the relative increases?
  
 With regard to accuracy, can anyone speak to what kind of situation
 might demonstrate the benefit of the new algorithm?  i.e. transcoding,
 SpanDSP, Analog interfaces (DAHDI) etc.
  
 Thanks
 Karl

Karl--

I was involved in merging those patches into Asterisk. They are slightly
slower than the original u/a-law algorithms, but not much slower. The
u/a-law code are the fastest codecs Asterisk has. As Kevin once said,
1.4 x of 0 is still zero. The author of the fixes told me that his
fixes straighten out problems with coding vs. decoding that were in
the original code. In the original code, he said, after 4 or so hops,
fax transmissions would no longer work.

Hope this helps.

murf


-- 
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Software Developer
Digium


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[asterisk-users] Request for testing of new driver for B410P Quad-Port BRI

[Shaun Ruffell]

There are new release candidates for dahdi-linux (2.1.0-rc3), dahdi-tools
(2.1.0-rc3), and dahdi-linux-complete (2.1.0-rc3+2.1.0-rc3, a combination of
dahdi-linux and dahdi-tools in one package) that contain a new DAHDI driver
for the B410P Quad-Port BRI card.

http://www.digium.com/en/products/digital/b410p.php

If you are a user of the B410P card, and are able, please test these release
candidates in your environment.  To test you will need version 1.4.4 or
greater of libpri and version 1.6.0 or greater of Asterisk.

You can download the dahdi-linux-complete release candidate at:
http://downloads.digium.com/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.1.0-rc3+2.1.0-rc3.tar.gz

or via svn from:
http://svn.digium.com/svn/dahdi/linux-complete/tags/2.1.0-rc3+2.1.0-rc3

The file system.conf.sample in dahdi-tools contains notes about configuring
DAHDI for the BRI card.

http://svn.digium.com/view/dahdi/tools/tags/2.1.0-rc3/system.conf.sampleview=markup

The ChangeLogs with all the changes for these releases are at:
http://svn.digium.com/svn/dahdi/linux/tags/2.1.0-rc3/ChangeLog
and
http://svn.digium.com/svn/dahdi/tools/tags/2.1.0-rc3/ChangeLog

Many Thanks,
Shaun

Shaun Ruffell
[EMAIL PROTECTED]


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[asterisk-users] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server

[Jason Lixfeld]

I'm having some issues getting app_voicemail_imapstorage to talk to my  
IMAP server.  From imapstorage.txt, I've got the voicemail.conf  
configured properly, but if I leave a voicemail for extension , I  
see no indication that the module is trying to reach the IMAP server.   
What am I missing?

# voicemail.conf
[general]
imapserver=172.16.17.2

[default]
 = ,,,,imapuser=joe|imappassword=joespassword

# full.log | grep -i voicemail
[Nov 11 18:11:37] VERBOSE[13681] logger.c: -- Reloading module  
'app_voicemail_imapstorage.so' (Comedian Mail (Voicemail System))
[Nov 11 18:11:37] VERBOSE[13681] logger.c:   == Parsing '/etc/asterisk/ 
voicemail.conf': [Nov 11 18:11:37] DEBUG[13681] config.c: Parsing /etc/ 
asterisk/voicemail.conf
[Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM  
Temperary Greeting Reminder Option disabled globally
[Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM CID  
Info before msg disabled globally
[Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Send  
Voicemail msg disabled globally
[Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: ENVELOPE  
before msg enabled globally
[Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Duration  
info before msg enabled globally
[Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] 
exten-vm:15] NoOp(SIP/-097abc68, Voicemail is ) in new stack
[Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] 
exten-vm:17] NoOp(SIP/-097abc68, Sending to Voicemail box  
) in new stack
[Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- 
[EMAIL PROTECTED]:1] NoOp(SIP/-097abc68, BUSY voicemail) in new  
stack
[Nov 11 18:12:14] DEBUG[13893] pbx.c: Launching 'VoiceMail'
[Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- 
[EMAIL PROTECTED]:3] VoiceMail(SIP/-097abc68, [EMAIL PROTECTED]|b) in  
new stack
[Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: Before  
find_user
[Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: /var/spool/ 
asterisk/voicemail/default//busy doesn't exist, doing what we can
[Nov 11 18:12:26] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ 
voicemail/default//INBOX'
[Nov 11 18:12:26] DEBUG[13893] app.c: Unlocked path '/var/spool/ 
asterisk/voicemail/default//INBOX'
[Nov 11 18:12:26] DEBUG[13893] app.c: play_and_record: None, /var/ 
spool/asterisk/voicemail/default//tmp/t4mOcQ, 'wav49|wav'
[Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=0, open writing:  / 
var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav49,  
0x9844ad0
[Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=1, open writing:  / 
var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav,  
0x98219f0
[Nov 11 18:12:33] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ 
voicemail/default//INBOX'
[Nov 11 18:12:33] DEBUG[13893] app.c: Unlocked path '/var/spool/ 
asterisk/voicemail/default//INBOX'


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Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

[Wilton Helm]

I'm a bit puzzled, also, having implemented ulaw and alaw in an embedded 
application.  Each can be done with a 16 Kbyte table in about 0 time with no 
errors.  There are probably tricks that will cut the table down by 2 or 4 X for 
a small cost in CPU cycles.  The inverse requires 256 16 bit words.  I thought 
ulaw and alaw were pretty much no brainers.  I don't know of any gottchas.  Why 
anyone with more that a few K bytes of total system memory would even consider 
anything other than a lookup table is beyond me.

Wilton
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Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

[Steve Murphy]

On Tue, 2008-11-11 at 16:11 -0700, Wilton Helm wrote:
 I'm a bit puzzled, also, having implemented ulaw and alaw in an
 embedded application.  Each can be done with a 16 Kbyte table in about
 0 time with no errors.  There are probably tricks that will cut the
 table down by 2 or 4 X for a small cost in CPU cycles.  The inverse
 requires 256 16 bit words.  I thought ulaw and alaw were pretty much
 no brainers.  I don't know of any gottchas.  Why anyone with more that
 a few K bytes of total system memory would even consider anything
 other than a lookup table is beyond me.
  
 Wilton

Wilton--

AFAIK, the current algorithms (old  new) are indeed table lookup.
It wouldn't hurt for you to do a code review on them, you might
be able to improve them...!

murf

  
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Re: [asterisk-users] AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server

[Jason Parker]

It apparently isn't built with IMAP support.  That would be a bug in my
packaging.  I'll see what I can do with it.

Jason Lixfeld wrote:
 I'm having some issues getting app_voicemail_imapstorage to talk to my  
 IMAP server.  From imapstorage.txt, I've got the voicemail.conf  
 configured properly, but if I leave a voicemail for extension , I  
 see no indication that the module is trying to reach the IMAP server.   
 What am I missing?
 
 # voicemail.conf
 [general]
 imapserver=172.16.17.2
 
 [default]
  = ,,,,imapuser=joe|imappassword=joespassword
 
 # full.log | grep -i voicemail
 [Nov 11 18:11:37] VERBOSE[13681] logger.c: -- Reloading module  
 'app_voicemail_imapstorage.so' (Comedian Mail (Voicemail System))
 [Nov 11 18:11:37] VERBOSE[13681] logger.c:   == Parsing '/etc/asterisk/ 
 voicemail.conf': [Nov 11 18:11:37] DEBUG[13681] config.c: Parsing /etc/ 
 asterisk/voicemail.conf
 [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM  
 Temperary Greeting Reminder Option disabled globally
 [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: VM CID  
 Info before msg disabled globally
 [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Send  
 Voicemail msg disabled globally
 [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: ENVELOPE  
 before msg enabled globally
 [Nov 11 18:11:37] DEBUG[13681] app_voicemail_imapstorage.c: Duration  
 info before msg enabled globally
 [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] 
 exten-vm:15] NoOp(SIP/-097abc68, Voicemail is ) in new stack
 [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [EMAIL PROTECTED] 
 exten-vm:17] NoOp(SIP/-097abc68, Sending to Voicemail box  
 ) in new stack
 [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- 
 [EMAIL PROTECTED]:1] NoOp(SIP/-097abc68, BUSY voicemail) in new  
 stack
 [Nov 11 18:12:14] DEBUG[13893] pbx.c: Launching 'VoiceMail'
 [Nov 11 18:12:14] VERBOSE[13893] logger.c: -- Executing [s- 
 [EMAIL PROTECTED]:3] VoiceMail(SIP/-097abc68, [EMAIL PROTECTED]|b) in 
  
 new stack
 [Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: Before  
 find_user
 [Nov 11 18:12:14] DEBUG[13893] app_voicemail_imapstorage.c: /var/spool/ 
 asterisk/voicemail/default//busy doesn't exist, doing what we can
 [Nov 11 18:12:26] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ 
 voicemail/default//INBOX'
 [Nov 11 18:12:26] DEBUG[13893] app.c: Unlocked path '/var/spool/ 
 asterisk/voicemail/default//INBOX'
 [Nov 11 18:12:26] DEBUG[13893] app.c: play_and_record: None, /var/ 
 spool/asterisk/voicemail/default//tmp/t4mOcQ, 'wav49|wav'
 [Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=0, open writing:  / 
 var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav49,  
 0x9844ad0
 [Nov 11 18:12:26] VERBOSE[13893] logger.c: -- x=1, open writing:  / 
 var/spool/asterisk/voicemail/default//tmp/t4mOcQ format: wav,  
 0x98219f0
 [Nov 11 18:12:33] DEBUG[13893] app.c: Locked path '/var/spool/asterisk/ 
 voicemail/default//INBOX'
 [Nov 11 18:12:33] DEBUG[13893] app.c: Unlocked path '/var/spool/ 
 asterisk/voicemail/default//INBOX'
 
 
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Re: [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?

[Jim Duda]

 When it fails, I get this message:
 [Nov  9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of 
 type 'DAHDI' (cause 0 - Unknown)
 
 Can you enable debug logging? Do you see any message about the casue for
 that?

Yes, I enabled logging, however, no additional logging was available.
I instrumented the code myself with additional logging.  I have provided the 
code snipet 
from chan_dahdi.c below.  It appears that my problem is caused by the TDM 
channel being
in the Onhook state.  Something makes the channel go Offhook, and things begin 
to work
properly.  I'm able to solve the Onhook case by changing the code to always 
return 1
when Onhook, indicating Offhook.

I'm hoping someone might shed some light on the onhook comments here.  What 
causes both
par.rxbits == -1 and par.rxisoffhook == 0?

chan_dahdi.c line 3786

if (res) {
ast_log(LOG_WARNING, Unable to check hook state on channel %d: %s\n, 
p-channel, strerror(errno));
} else if ((p-sig == SIG_FXSKS) || (p-sig == SIG_FXSGS)) {
/* When onhook that means no battery on the line, and thus
  it is out of service..., if it's on a TDM card... If it's a channel
  bank, there is no telling... */
if (par.rxbits  -1)
return 1;
if (par.rxisoffhook)
return 1;
else {
ast_log(LOG_WARNING, available 6c par.rxbits: %d par.rxisoffhook: 
%d\n, par.rxbits, par.rxisoffhook);
//return 0;
return 1;
}

Jim



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Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

[Steve Underwood]

Wilton Helm wrote:
 I'm a bit puzzled, also, having implemented ulaw and alaw in an 
 embedded application.  Each can be done with a 16 Kbyte table in about 
 0 time with no errors.  There are probably tricks that will cut the 
 table down by 2 or 4 X for a small cost in CPU cycles.  The inverse 
 requires 256 16 bit words.  I thought ulaw and alaw were pretty much 
 no brainers.  I don't know of any gottchas.  Why anyone with more that 
 a few K bytes of total system memory would even consider anything 
 other than a lookup table is beyond me.
Lookup tables occupy cache, though with the latest CPUs getting 12M of 
cache a couple of 16k byte tables may not be an issue. When the cache 
was 256k it was.

What is puzzling here is why the new code should be slower. Asterisk has 
always implemented these codecs incorrectly. In fact, a lot of projects 
did. They all used the same broken 1 page implementation from Sun :-). 
Most projects sorted this out some time ago, but the Asterisk people 
didn't seem to care. Even with a single transcoding the Sun A-law code 
sounds bad on quiet voices. Nobody else saw a speed drop when they 
changed. In fact, the only way to do a-law - u-law correctly is with a 
256 byte lookup tables, so that *has* to be fast.

Steve


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[asterisk-users] Grandstream and pickup

[Julian Lyndon-Smith]

Man, I really feel stupid, but after banging my head on a brick wall for 
several hours ... I need help!

I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 
5707, and I've got an xlite on 5608.

When I make a call from an outside line, I dial SIP/5608. The little 
blinky light on the GXP that's monitoring 5608 goes, well, blink 
blink. :) I then press the button and get a 603 declined. This is the 
dialplan smippet:

[blf]
exten = 5608,hint,SIP/5608
..snip..

..snip..
exten =  444608,1,Dial(Sip/5608)
..snip..

exten = _**,1,Pickup(${EXTEN:2})
exten = _**,n,Hangup()

both phones are in the same context, callgroup and pickupgroup. I've 
tried adding all of the @context that I have, to no avail.

Accepting call from 'xx' to 'xxx608' on channel 0/1, span 1
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, SIP/5608) in new 
stack
 Extension Changed 5608[blf] new state Ringing for Notify User 5707
-- Called 5608
-- SIP/5608-083ea7a8 is ringing
 Extension Changed 5707[blf] new state InUse for Notify User 5707
-- Executing [EMAIL PROTECTED]:1] Pickup(SIP/5707-08393e10, 5608) 
in new stack
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/5707-08393e10, ) in 
new stack
  == Spawn extension (from-sip, **5608, 2) exited non-zero on 
'SIP/5707-08393e10'
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/5707-08393e10,  HANGUP 
FROM-SIP  [/16]) in new stack
 Extension Changed 5707[blf] new state Idle for Notify User 5707 (queued)
 Extension Changed 5707[blf] new state Idle for Notify User 5707
-- Channel 0/1, span 1 got hangup request, cause 16
Extension Changed 5608[blf] new state Idle for Notify User 5707

Any help would be much appreciated.

Thanks

Julian


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Re: [asterisk-users] OT: Polycom Firmware available (by accident?)

[Darryl Dunkin]

Instead, they are likely releasing something newer and better. I believe
they have always had SIP software for download, however, it is never the
most recent. They only provide 'previous software' for end-users, if you
want the latest, you still have to go to your vendor.

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip_
upgrade.html

There are some direct download links to the previous versions here,
which are often newer than what is listed on the product support page
(such as the 501s, only list something around 2.1.2):
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Fullerton
Sent: Tuesday, November 11, 2008 08:50
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] OT: Polycom Firmware available (by
accident?)

Jared Smith wrote:
 On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
 Anyone looking for firmware should get it now before it disappears.
 
 It's my understanding that this isn't a fluke, but that Polycom has
 indeed changed their policy and will no longer you to go through your
 reseller to get the latest and greatest firmware.
 

That's awesome!

I had wondered that but since I hadn't seen links for 3.1.0B or the new 
BootROM's it made me a little suspicious.

Thanks for the clarification.

-Dave

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Re: [asterisk-users] Grandstream and pickup

[Julian Lyndon-Smith]

A little more information:

If I change the dial command to

..snip..
exten =  444608,1,Set(__PICKUPMARK=5608)
exten =  444608,n,Dial(Sip/5608)
..snip..

and the pickup command to

exten = _**,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**,n,Hangup()

then it works ...

Julian

Julian Lyndon-Smith wrote:
 Man, I really feel stupid, but after banging my head on a brick wall for 
 several hours ... I need help!

 I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 
 5707, and I've got an xlite on 5608.

 When I make a call from an outside line, I dial SIP/5608. The little 
 blinky light on the GXP that's monitoring 5608 goes, well, blink 
 blink. :) I then press the button and get a 603 declined. This is the 
 dialplan smippet:

 [blf]
 exten = 5608,hint,SIP/5608
 ..snip..

 ..snip..
 exten =  444608,1,Dial(Sip/5608)
 ..snip..

 exten = _**,1,Pickup(${EXTEN:2})
 exten = _**,n,Hangup()

 both phones are in the same context, callgroup and pickupgroup. I've 
 tried adding all of the @context that I have, to no avail.

 Accepting call from 'xx' to 'xxx608' on channel 0/1, span 1
 -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, SIP/5608) in new 
 stack
  Extension Changed 5608[blf] new state Ringing for Notify User 5707
 -- Called 5608
 -- SIP/5608-083ea7a8 is ringing
  Extension Changed 5707[blf] new state InUse for Notify User 5707
 -- Executing [EMAIL PROTECTED]:1] Pickup(SIP/5707-08393e10, 5608) 
 in new stack
 -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/5707-08393e10, ) in 
 new stack
   == Spawn extension (from-sip, **5608, 2) exited non-zero on 
 'SIP/5707-08393e10'
 -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/5707-08393e10,  HANGUP 
 FROM-SIP  [/16]) in new stack
  Extension Changed 5707[blf] new state Idle for Notify User 5707 (queued)
  Extension Changed 5707[blf] new state Idle for Notify User 5707
 -- Channel 0/1, span 1 got hangup request, cause 16
 Extension Changed 5608[blf] new state Idle for Notify User 5707

 Any help would be much appreciated.

 Thanks

 Julian


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[asterisk-users] AS5200 - T100P - No alarms but no calls either...

[Don Fanning]

Greetings,

I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover 
cable.  I got it where the alarms are all ok/green but I'm unable to 
dial out or dial into the AS5200.

Anyone have any suggestions as to where to begin troubleshooting this?

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Re: [asterisk-users] DNS A queries for channel

[Matt Riddell]

On 12/11/2008 6:20 a.m., Tilghman Lesher wrote:
 On Tuesday 11 November 2008 05:17:30 Matt Riddell wrote:
 On 11/11/2008 10:48 p.m., samuel wrote:
 So far I've updated a few machines (1.4.22) and the DNS queries are
 reduced to a minimum, at least haven't seen DNS channel queries...
 Slightly more useful to me, does anyone know the patch that fixed it, so
 I can update machines I don't want to upgrade to 1.4.22?

 http://svn.digium.com/view/asterisk?view=revrevision=133649

Thanks for that man.

-- 
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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[asterisk-users] Use DECT GAP handsets with Snom M3 base?

[Paul Chambers]

Anyone have practical experience using inexpensive GAP-compliant DECT 
handsets with the Snom M3 basestation?

When I asked Snom support, the answer was that 'basic functionality 
should work', but they didn't elaborate. I'm _guessing_ that means 
registering/unregistering with the base, making calls, and receiving 
calls (including presenting caller ID). They also stated that they did 
not test with other manufacturer's products (implying 'you're on your 
own'). Pretty much the response I expected.

This would be SOHO/home use. We're using the 'ATA + multi-handset 
cordless phone' approach right now, but there are definitely advantages 
to having Asterisk see each handset as an extension.

Looks like the Snom handsets are running at least US$130 apiece, so it'd 
be an expensive proposition to replace all our handsets. By comparison, 
a mainstream Panasonic DECT 6.0 handset runs ~US$30 (and I already have 
six...).  I'm thinking a couple of 'nice' M3 handsets, and low-cost ones 
for the rest of the house. It's not a big deal if a $30 handset gets 
trashed, but $130 is a different story.

Appreciate any experiences you can share,

-- Paul

p.s. I'm in the U.S.

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Re: [asterisk-users] Use DECT GAP handsets with Snom M3 base?

[Michael Graves]

On Tue, 11 Nov 2008 18:26:01 -0800, Paul Chambers wrote:

Anyone have practical experience using inexpensive GAP-compliant DECT 
handsets with the Snom M3 basestation?

When I asked Snom support, the answer was that 'basic functionality 
should work', but they didn't elaborate. I'm _guessing_ that means 
registering/unregistering with the base, making calls, and receiving 
calls (including presenting caller ID). They also stated that they did 
not test with other manufacturer's products (implying 'you're on your 
own'). Pretty much the response I expected.

This would be SOHO/home use. We're using the 'ATA + multi-handset 
cordless phone' approach right now, but there are definitely advantages 
to having Asterisk see each handset as an extension.

Looks like the Snom handsets are running at least US$130 apiece, so it'd 
be an expensive proposition to replace all our handsets. By comparison, 
a mainstream Panasonic DECT 6.0 handset runs ~US$30 (and I already have 
six...).  I'm thinking a couple of 'nice' M3 handsets, and low-cost ones 
for the rest of the house. It's not a big deal if a $30 handset gets 
trashed, but $130 is a different story.

Appreciate any experiences you can share,

Paul,

I have the snom m3 system with two handsets at the moment. My wife
would take issue with your characterisation of the snom handsets as
nice. She thinks that they're too small and the smooth buttons are a
bother. I have no such reservations. I generally like them.

I have used the m3 side-by-side with a Siemens S685IP. Both claim to be
GAP compliant and yet they worked as totally separate systems. I might
have expected some interaction. Perhaps the snom handset attempting to
register with the siemens based or vice versa. But that did not happen.

There is a well qualified rumour that Siemens will soon be offering
their DECT wares in the US. This is very interesting as some of these
are G.722 capable, although they have some quirks as well. Like no mic
mute button for example. They are cheaper than the snom systems.

Finally, Aastra is about to release their MBU-400, which is a SIP/DECT
system from RTX Telecom...just like snom's m3. It appears to be on move
up the hardware ladder, supporting one POTS line as well as 8 ITSP
accounts. But it's even more expensive than the snom.

Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245




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Re: [asterisk-users] set(CALLERID(name) not working

[sean darcy]

C F wrote:
 Who you calling? Is it a remote non PSTN phone number? Or a PSTN number?
 
It's incoming. Both pstn and voip.

sean


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Re: [asterisk-users] set(CALLERID(name) not working

[sean darcy]

sean darcy wrote:
 I've tried to create a subroutine that sets callerid name based on number.
 
 extensions.conf:
 
 ...
 exten = s,1,Answer()
 exten = s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
 exten = s,n,Dial(${mainline},60)
 ...
 
 [set-callerid-name]
 exten = 0,1,NoOp( no CALLERID num set)
 exten = 02025462677,1,Set(CALLERID(name) = Fred )
 
 exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
 exten = _X.,3,Return()
 
 But it doesn't work. CALLERID(name) isn't changed:
 
 ...
  -- Executing [EMAIL PROTECTED]:2] Answer(IAX2/john-7775, ) 
 in new stack
  -- Executing [EMAIL PROTECTED]:4] Gosub(IAX2/john-7775, 
 set-callerid-name|02025462677|1) in new stack
  -- Executing [EMAIL PROTECTED]:1] 
 Set(IAX2/john-7775, CALLERID(name) = Fred ) in new stack
  -- Executing [EMAIL PROTECTED]:2] 
 NoOp(IAX2/john-7775, CALLERID: Cell Phone   CT) in new stack
  -- Executing [EMAIL PROTECTED]:3] 
 Return(IAX2/john-7775, ) in new stack
  -- Executing [EMAIL PROTECTED]:5] Dial(IAX2/john-7775, 
 DAHDI/1|60) in new stack
  -- Called 1
 
 
 
 Why not? How come CALLERID(name) isn't Fred??
 
 sean
 
 

Well I never did find out. What I did find out is that I could only set 
CALLERID(name) in the context the received the call, not in the 
subroutine. But if I set a dummy variable ( Set(cidname=Fred) ) in the 
subroutine, I could set the callerid back in the originating context ( 
Set(CALLERID(name) = ${cidname} ).

Makes no sense, but there you are.

sean


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Re: [asterisk-users] set(CALLERID(name) not working

[Daniel Lynes]

You'll need to lose the double quotation marks in the assignment:

Set(CALLERID(name)=Fred) becomes:
Set(CALLERID(name)=Fred)

If it still doesn't work, then it means that your particular provider 
does not support the ability to be able to set the caller ID name, or 
it's receiving a corrupted copy of it.  One such provider that I'm aware 
of for that issue is Group Telecom.  They receive corrupted caller ID 
name information from asterisk.  So, caller ID num information will 
work, but not caller ID name information.


sean darcy wrote:

sean darcy wrote:
  

I've tried to create a subroutine that sets callerid name based on number.

extensions.conf:

...
exten = s,1,Answer()
exten = s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
exten = s,n,Dial(${mainline},60)
...

[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )

exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
exten = _X.,3,Return()



--

//
//  Daniel Bruce Lynes  //
//  Westwood Village Computers  //
//  http://www.westwoodvillagecomputers.com///
//  //
//  Opinions expressed are not necessarily those of //
//  Westwood Village Computers. //
//

begin:vcard
fn:Daniel Lynes
n:Lynes;Daniel
org:Westwood Village Computers
adr:;;;Coquitlam;BC;V3B 0B2;Canada
email;internet:[EMAIL PROTECTED]
title:Owner
tel;work:604-484-0151
tel;cell:604-728-3777
url:http://www.westwoodvillagecomputers.com/
version:2.1
end:vcard

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Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

[Wilton Helm]

It wouldn't hurt for you to do a code review on them,

I'd better get more up to speed on * in general first.  It would be interesting 
to compare them to my code.  However, I don't have a useful * installation 
here, yet--I'm working on it.

Wilton


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Re: [asterisk-users] What makes TDM400 FXS Connection to TELCO go into Off Hook State?

[Tzafrir Cohen]

On Tue, Nov 11, 2008 at 07:05:23PM -0500, Jim Duda wrote:
 
  When it fails, I get this message:
  [Nov  9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of 
  type 'DAHDI' (cause 0 - Unknown)
  
  Can you enable debug logging? Do you see any message about the casue for
  that?
 
 Yes, I enabled logging, however, no additional logging was available.
 I instrumented the code myself with additional logging.  I have provided the 
 code snipet 
 from chan_dahdi.c below.  It appears that my problem is caused by the TDM 
 channel being
 in the Onhook state.  Something makes the channel go Offhook, and things 
 begin to work
 properly.  I'm able to solve the Onhook case by changing the code to always 
 return 1
 when Onhook, indicating Offhook.
 
 I'm hoping someone might shed some light on the onhook comments here.  What 
 causes both
 par.rxbits == -1 and par.rxisoffhook == 0?
 
 chan_dahdi.c line 3786
 
 if (res) {
   ast_log(LOG_WARNING, Unable to check hook state on channel %d: %s\n, 
 p-channel, strerror(errno));
 } else if ((p-sig == SIG_FXSKS) || (p-sig == SIG_FXSGS)) {
   /* When onhook that means no battery on the line, and thus
 it is out of service..., if it's on a TDM card... If it's a channel
 bank, there is no telling... */
   if (par.rxbits  -1)
   return 1;
   if (par.rxisoffhook)
   return 1;
   else {
   ast_log(LOG_WARNING, available 6c par.rxbits: %d par.rxisoffhook: 
 %d\n, par.rxbits, par.rxisoffhook);
 //return 0;
   return 1;
   }

Interesting. This part was originally ifdef-ed out in chan_zap:
http://bugs.digium.com/13786

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] AS5200 - T100P - No alarms but no calls either...

[Tzafrir Cohen]

On Tue, Nov 11, 2008 at 06:02:49PM -0800, Don Fanning wrote:
 Greetings,
 
 I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover 
 cable.  I got it where the alarms are all ok/green but I'm unable to 
 dial out or dial into the AS5200.
 
 Anyone have any suggestions as to where to begin troubleshooting this?

pri show span 1

pri debug span 1

And then see what happens on a call.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] play file from url

[Pezhman Lali]

mp3player, is just for your need,

use it this like

exten = _X.,1,mp3player(http://www.test.com/test.mp3;)

try this page
http://www.voip-info.org/wiki-Asterisk+cmd+MP3Player
best

--- On Wed, 11/12/08, Singer X.J. Wang [EMAIL PROTECTED] wrote:
From: Singer X.J. Wang [EMAIL PROTECTED]
Subject: Re: [asterisk-users] play file from url
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Wednesday, November 12, 2008, 2:00 AM




  
Mike Clark wrote:

  I would like to do something like:

exten = s,1,playback(http://my.server.com/file.wav)

I tested and it does not work. It seems highly likely that someone would 
already have done this one way or another. I know I could do a system 
wget and then play the local file, but wanted something a bit more elegant.

Thanks,

Mike Clark

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One issue with that is latency... 



-- 

Singer Wang

System and Database Engineer

The Pythian Group




  

  Office: 
   
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Re: [asterisk-users] Grandstream and pickup

[Julian Lyndon-Smith]

Arrgh. this is driving me nuts. Can anyone put me out of my misery ? 
Pretty please ;)

Julian Lyndon-Smith wrote:
 Man, I really feel stupid, but after banging my head on a brick wall for 
 several hours ... I need help!

 I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 
 5707, and I've got an xlite on 5608.

 When I make a call from an outside line, I dial SIP/5608. The little 
 blinky light on the GXP that's monitoring 5608 goes, well, blink 
 blink. :) I then press the button and get a 603 declined. This is the 
 dialplan smippet:

 [blf]
 exten = 5608,hint,SIP/5608
 ..snip..

 ..snip..
 exten =  444608,1,Dial(Sip/5608)
 ..snip..

 exten = _**,1,Pickup(${EXTEN:2})
 exten = _**,n,Hangup()

 both phones are in the same context, callgroup and pickupgroup. I've 
 tried adding all of the @context that I have, to no avail.

 Accepting call from 'xx' to 'xxx608' on channel 0/1, span 1
 -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, SIP/5608) in new 
 stack
  Extension Changed 5608[blf] new state Ringing for Notify User 5707
 -- Called 5608
 -- SIP/5608-083ea7a8 is ringing
  Extension Changed 5707[blf] new state InUse for Notify User 5707
 -- Executing [EMAIL PROTECTED]:1] Pickup(SIP/5707-08393e10, 5608) 
 in new stack
 -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/5707-08393e10, ) in 
 new stack
   == Spawn extension (from-sip, **5608, 2) exited non-zero on 
 'SIP/5707-08393e10'
 -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/5707-08393e10,  HANGUP 
 FROM-SIP  [/16]) in new stack
  Extension Changed 5707[blf] new state Idle for Notify User 5707 (queued)
  Extension Changed 5707[blf] new state Idle for Notify User 5707
 -- Channel 0/1, span 1 got hangup request, cause 16
 Extension Changed 5608[blf] new state Idle for Notify User 5707

 Any help would be much appreciated.

 Thanks

 Julian


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Re: [asterisk-users] Grandstream and pickup

[Gordon Henderson]

On Wed, 12 Nov 2008, Julian Lyndon-Smith wrote:

 Man, I really feel stupid, but after banging my head on a brick wall for
 several hours ... I need help!

AIUI - Pickup works on an extension..

So if the xlite is SIP/5608, but extension is 444608, then you need to 
pickup 444608.

Gordon


 I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten
 5707, and I've got an xlite on 5608.

 When I make a call from an outside line, I dial SIP/5608. The little
 blinky light on the GXP that's monitoring 5608 goes, well, blink
 blink. :) I then press the button and get a 603 declined. This is the
 dialplan smippet:

 [blf]
 exten = 5608,hint,SIP/5608
 ..snip..

 ..snip..
 exten =  444608,1,Dial(Sip/5608)
 ..snip..

 exten = _**,1,Pickup(${EXTEN:2})
 exten = _**,n,Hangup()

 both phones are in the same context, callgroup and pickupgroup. I've
 tried adding all of the @context that I have, to no avail.

 Accepting call from 'xx' to 'xxx608' on channel 0/1, span 1
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, SIP/5608) in new
 stack
 Extension Changed 5608[blf] new state Ringing for Notify User 5707
-- Called 5608
-- SIP/5608-083ea7a8 is ringing
 Extension Changed 5707[blf] new state InUse for Notify User 5707
-- Executing [EMAIL PROTECTED]:1] Pickup(SIP/5707-08393e10, 5608)
 in new stack
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/5707-08393e10, ) in
 new stack
  == Spawn extension (from-sip, **5608, 2) exited non-zero on
 'SIP/5707-08393e10'
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/5707-08393e10,  HANGUP
 FROM-SIP  [/16]) in new stack
 Extension Changed 5707[blf] new state Idle for Notify User 5707 (queued)
 Extension Changed 5707[blf] new state Idle for Notify User 5707
-- Channel 0/1, span 1 got hangup request, cause 16
 Extension Changed 5608[blf] new state Idle for Notify User 5707

 Any help would be much appreciated.

 Thanks

 Julian


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