date:20080904

 I have a Grandstream GXP1200 and eager to try this codec.  I've heard
 good things about the quality.

 Anyone tried it with asterisk?

 I can't until 1.6 is released.


 I have used G.722 with Asterisk many times.  If you have more specific
 questions about it and Asterisk, I would be happy to try to answer them.

Specifically my questions are,

[1] The quality of voice between g722 and say GSM or 729
[2] Interoperability between phones with g722 and other codecs
[3] Asterisk support for G722 phones.

Thanks!
Steve

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Re: [asterisk-users] DID number

2008-09-04 Thread michel freiha

Dear Sir,

Please find below the error that we are getting when enabling 'sip set
debug'.


localhost*CLI
--- Reliably Transmitting (no NAT) to 83.202.82.39:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 83.202.82.39:5060
;branch=z9hG4bK5ac79f249887f915005b5d34415b1a56;received=83.202.82.39
From: 96155 sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
;tag=48201
To: sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
;tag=as3fc2e680
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0



[Sep  4 11:13:05] NOTICE[10388]: chan_sip.c:14035 handle_request_invite:
Call from 'sip_proxy' to extension 'DID_Number' rejected because extension
not found.
Scheduling destruction of SIP dialog
'[EMAIL PROTECTED]' in 32000 ms (Method: INVITE)
localhost*CLI
--- SIP read from 83.202.82.39:5060 ---
ACK sip:[EMAIL PROTECTED] [EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
From: 96155 sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
;tag=48201
To: sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
;tag=as3fc2e680
Via: SIP/2.0/UDP 83.202.82.39:5060
;branch=z9hG4bK5ac79f249887f915005b5d34415b1a56
Max-Forwards: 69
User-Agent: Vox Callcontrol
Content-Length: 0



Regards



On 9/4/08, Jaswinder Singh [EMAIL PROTECTED] wrote:

 [442033553]
 user=442033553
 type=pusers
 secret=1234
 host=dynamic
 context=users
 nat=yes


 make it context=stations , i am assuming this is how your DID provider
 is sending u calls ?

 Let us know if your DID provider is just sending calls to your ip
 address or you are registering asterisk server with the, . Keep
 context=stations in extensions.conf  global section .

 On Thu, Sep 4, 2008 at 2:41 AM, Igor Hernandez [EMAIL PROTECTED] wrote:
  Hey,
 
  Did you reload asterisk after changing the extensions.conf?
 
  Also, if you try it with sip set debug on the console what do you see?
 
 
  michel freiha wrote:
  Hello Air,
 
  I did what you asked for but I got the following error:
 
  extensions.conf:
 
  [stations]
  exten = 442033553,1,Answer
  exten = 442033553,n,Playback(demo-nogo)
 
  Error message:
  [Sep  3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite:
  Call from '' to extension '442033553' rejected because extension not
 found.
  Regards
  On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] wrote:
 
  michel freiha wrote:
   Hi All,
   I bought a DID number from VOxbone...this number could be dialed
 from
   any PSTN line and could be forwarded to any SIP server like
 asterisk
   server...Now I need to forward this number to my asterisk server
  so when
   a customer dial this number from his GSM or Land line PSTN number
 the
   call will be forwarde to my asterisk server and I need to play a
 wav
   file for example..
   Can you please give me some tips about how to accomplish this
 task?
  
   Regards
  
  
  
 
 
  
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  Hello,
 
  I have never used that provider but usually either the provider
 knows
  your switch's ip and routes the did traffic to it or you have
 asterisk
  register with the provider so that it knows where to route the
 calls.
 
  Once thats done you can do something like
 
  exten = XX,1,Answer
  exten = XX,n,Playback(file)
 
  Where the x's are the number that you see coming in from your
 provider.
  If you're routed all your dids from what looks like one
  number(callcentric does this) then you might need to use the sip
 header
  to route your did to the particular extension you want. You
 shouldn't
  have to bother with this if you only have one did.
 
 
  Regards,
 
  --
  Igor Hernandez
  Escape Communications
  http://www.escapetel.com http://www.escapetel.com/
 
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Re: [asterisk-users] dial out via fxo gateway

On Thu, Sep 4, 2008 at 4:44 PM, ACL [EMAIL PROTECTED] wrote:
 My current config:

 pstn - audiocodes fxo gateway - asterisk - xlite

 every fxo ports are registered with asterisk

 I have this extensions.conf

 exten = 111,1,answer
 exten = 111,n,dial(sip/fxo1)
 exten = 111,n,hangup

 If we dial 111 by xlite, I could hear pstn dialing tone. I could key in a
 phone no and connect to the called party. this is a two stage dialing.

 How could we preset a phone no. in the extensions.conf without having the
 sip client keys in the phone no (ONE STAGE DIALING)? I do not want to preset
 the phone no. in fxo gateway.  the phone no. must be modifiable.

 pls kindly advise.


I usually have a simple outbound context

[outbound]
exten = _9X.,1,Dial(Zap/g0/${EXTEN:1})
exten = _9X.,n,Congestion()
exten = _9X.,n,Hangup()

Be warned that the above dialplan will allow calls with anykind of
numbers (even international). So be sure to pattern match depending of
where the calls should go.

Don't forget to include the [outbound] context in whatever context
your SIP extention is in.

Steve

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[asterisk-users] MixMonitor + Originate

2008-09-04 Thread Geraint Lee

Hi everyone,

I'm trying to get calls to record with the following setup:
Using phpagi originate is called from a web application:
$asm-originate(Local/ . $row['extension'] . @sip-standard,
$row['phone_number'], sip-standard, 1, , , 7000);

The agent being called is extension Local/[EMAIL PROTECTED] and the number
originated for the agent is [EMAIL PROTECTED]

MixMonitor records fine up until 100 answers then the recording stops, but
the CLI output suggests that the call is still being recorded...

extensions.conf and CLI output below...

Anyone have any ideas?


extensions.conf:
exten = 100,1,MixMonitor(test.wav)
exten = 100,2,Dial(SIP/${EXTEN})

exten = _1XX,1,Dial(SIP/${EXTEN})

Output from CLI:
  == Manager 'amis' logged on from 192.168.0.180
-- Executing [EMAIL PROTECTED]:1] Dial(Local/[EMAIL PROTECTED],2,
SIP/101) in new stack
-- Called 101
-- SIP/101-096f7ff8 is ringing
-- SIP/101-096f7ff8 answered Local/[EMAIL PROTECTED],2
-- Executing [EMAIL PROTECTED]:1]
MixMonitor(Local/[EMAIL PROTECTED],1, test.wav) in new stack
  == Begin MixMonitor Recording Local/[EMAIL PROTECTED],1
-- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],1,
SIP/100) in new stack
-- Called 100
-- Local/[EMAIL PROTECTED],1 requested special control 20, passing
it to SIP/100-09706218
  == Manager 'amis' logged off from 192.168.0.180
  == Spawn extension (sip-standard, 101, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
-- SIP/100-09706218 is ringing
-- SIP/100-09706218 answered SIP/101-096f7ff8
  == Spawn extension (sip-standard, 100, 3) exited non-zero on
'SIP/101-096f7ff8'
  == End MixMonitor Recording Local/[EMAIL PROTECTED],1

Cheers

Geraint
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Re: [asterisk-users] How to check mailbox exists (Received SIP subscribe for peer without mailbox)

2008-09-04 Thread Olivier

2008/9/4 Rob Hillis [EMAIL PROTECTED]

 Olivier wrote:
  Hi,
 
  I'm receiving this :
  [Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for
  peer without mailbox: 9163
 
  I've read this :
  http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html
 
  I typed this:
  asterisk -rx reload
  asterisk -rx voicemail show users
 
  ... and got :
  default9163  john doe 0
 
  This default ... line is 100% similar to others.

 Chances are that the mailbox line in sip.conf for this extension doesn't
 include the correct mailbox context.  Make sure that
 [EMAIL PROTECTED] is in the section of sip.conf for this extension
 and reload.


Hi,

You're right :  few minutes after posting this question, I saw  that
mailbox=9163 line was missing and that adding this line solved this.

Anyway, my 1st question remains as voicemail show users was somehow
misleading for me :
- on one hand, I've a notice telling Received SIP subscribe for peer
without mailbox: 9163
- on the other hand, voicemail show users shows something I understood as
mailbox exists.

Now that root cause is found, would you say that warnings or CLI should have
been different ?

Obviously, MWI subscriptions must come from SIP hardphones (at least those
supporting MWI feature).
So in this case, Received SIP subscribe for peer without mailbox: 9163
rather means Asterisk is receiving from SIP/9163, subscriptions to MWI, but
nothing in peer SIP/9163 settings is describing which mailbox should be the
scope of such subscriptions.

Do you agree ?




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Re: [asterisk-users] Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone

On Thu, 04 Sep 2008 00:42:21 -0500, Karl Fife wrote:

Has anyone seen or done an XML phone application integration using
Z-Wave or Zigbee (or legacy Crestron) for Office or Home automation,
lighting  thermostatic control, alarm systems etc?

If you've seen Crestron systems you may know that they are VERY
expensive, and proprietary control panels (some with color screens) and
special wiring add up to many tens of thousands of dollars.  Times are
changing, and it's now becoming obvious that this would be MUCH easier,
cheaper, more flexible, and reconfigurable etc using XML browser screen
phones back to a control interface on the MESH network using technolgies
like Zigbee  Z-Wave (or Crestron if already present).  You could even
integrate a control IVR using DISA.  Press 1 to ask Alison what's
turned on.  If you're doing thermostatic control, you could Press 2 to
ask Alison what's hot or not :-)

Does anyone have any experience with an integration like this?  Care to
share your story, or point us to your blog?

Thanks
-Karl

p.s.
Here's a possible programmable interface for programmatic control:
http://www.hawkingtech.com/products/productlist.php?CatID=43FamID=119ProdID=392
Using it may be a kludge.  No idea.  

I'm very inerested in this but I'm not settled upon which sort of
control hardware to use. I have some existing X-10 hardware which can
be controlled with simple serial commands, although its not fast or
especially reliable. All I really need is to turn on/off a few lights
and control some relays to open/close doors.

Also, I was thinking abotu control via the XML browser on the phone
initially. Possibly IVR based control later on.

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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[asterisk-users] New Install using DAHDI

2008-09-04 Thread Joseph L. Casale

I am about to setup a new Asterisk box which only uses SIP.
I used to simply use menuselect with Zaptel and choose the tools
that Asterisk required to exist and ztdummy.

Now with Dahdi, I am reading 
http://svn.digium.com/view/dahdi/tools/tags/2.0.0-rc2/UPGRADE.txt?view=co
and I understand I no longer can use menuselect as everything gets built.

Everything looks pretty trivial except the echo canceller portion.
I have never configured this on my SIP only systems without any
physical hardware (no digium cards) so can what should be my
strategy for running dahdi_cfg tool in this scenario where I
have a few SIP did's coming in with several SIP phones on the inside?

Thanks!
jlc

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[asterisk-users] Logs: messages, events, queue

2008-09-04 Thread bilal ghayyad

Hi List;

About logs existed under the folder /var/log/asterisk/, I would like to know 
the following:

1) How to enable/disable the messages log?
2) When messages log happen? Based on error or running application?
3) What difference between messages log and even log?
4) queue_log to be used for call center and queuing the call?

Any help?
Regards
Bilal


  

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Re: [asterisk-users] G722 and Asterisk 1.6

2008-09-04 Thread Russell Bryant


On Sep 4, 2008, at 6:25 AM, Steve Repo wrote:
 Specifically my questions are,

 [1] The quality of voice between g722 and say GSM or 729

I suppose that it's sort of subjective, but I think it sounds  
_awesome_.  It's a huge difference in quality to me.  You just need to  
try it out.  :)

 [2] Interoperability between phones with g722 and other codecs

Asterisk 1.4 does not support transcoding of G722.  Asterisk 1.6 does  
support transcoding, so it can be used in combination with any other  
codec that Asterisk supports.


 [3] Asterisk support for G722 phones.


Asterisk should work fine with any phone that supports that codec.   
Personally, I have only used it with Polycom phones.  Also, again,  
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has  
full support.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.





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Re: [asterisk-users] Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions)

2008-09-04 Thread Russell Bryant


On Sep 4, 2008, at 4:12 AM, z_gringo wrote:

 I have several asterisk servers running a couple of different  
 versions of 1.4.  One of our severs in California is running  
 1.4.18 with the Dial Plan in Realtime mySQL.  This server is storing  
 voicemails in the database connecting via odbc.  There are  
 approximately 900 sip users registered at any given time.   All of  
 the SIP users are in the sip.conf file, which is extracted from the  
 database.   Any time this server gets to around 90 simutaneous calls  
 (180 channels), the server is completely unstable.  On some  
 occasions, the asterisk process has continued to run, but is not  
 processing any calls or registrations.  On most occasions, the  
 asterisk process crashes, restarts, crashes again, etc.   During  
 periods of lower traffic, the system appears to be stable.

 Going back to version 1.4.8 or 1.4.11 seems to be stable, but there  
 is clearly a problem with 1.4.18 in this particular configuration.   
 The OS is 64 bit debian.   Has anyone seen something similar?

We would be happy to help figure out what's going wrong on your  
system.  However, the first step will have to be running the latest  
version.  So, please give 1.4.22 a try.  Then, please gather details  
and post them to http://bugs.digium.com/.  If you'd like to discuss  
what you need to do to create the bug report, join #asterisk-bugs on  
the freenode IRC network.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.





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[asterisk-users] conf files for dahdi

2008-09-04 Thread sean darcy

upgrading from zaptel to dahdi, with a TDM400P:

Is /etc/dahdi/system.conf the same as /etc/zaptel.conf? As I read the 
system.conf.sample, no echo canceller need be specified if there's a 
hardware ec. Can I just rename zaptel.conf?

What about zapata.conf? Is this just renamed 
/etc/asterisk/chan_dahdi.conf? Or zapata-channels.conf? Or just left alone?

sean


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Re: [asterisk-users] Dial L( x [: y ][: z ]) option truncates colon (:) using AGI /_

On Thursday 04 September 2008 03:15:51 selmak se wrote:
 AGI Rx  EXEC Dial SIP/1001,,L(32000:2:1)

 [Sep  4 11:04:20] WARNING[18100]: chan_sip.c:2907 create_addr: No such
 host: 1001,,L(32000

The issue is that internally, the application argument delimiter in 1.4 is
actually the pipe symbol, not the comma, so you'd need to do:
EXEC Dial SIP/1001||L(32000:2:1)

I agree that this is nonobvious, difficult to foresee, stupid, etc., which is
why starting in 1.6, the comma will become the application argument delimiter,
so your first form will work.  There is a backwards-compatibility option with
respect to using pipes that will enable a smooth transition for people used to
using the pipe as the delimiter.

-- 
Tilghman

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Re: [asterisk-users] #include changes in 1.4

On Thursday 04 September 2008 04:25:27 Chris Bagnall wrote:
 I finally got round to upgrading a few of our 1.2 servers to 1.4 over the
 last few days. Most of the changes in config files went without a hitch,
 but this one bothers me:

 ERROR[15836]: config.c:750 process_text_line: Future versions of Asterisk
 will treat a #include of a file that does not exist as an error, and will
 fail to load that configuration file.  Please ensure that the file
 '*/iax.conf' exists, even if it is empty.

 The iax.conf on these servers contains general parameters common to all
 users who connect, with the following line at the bottom:

 #include */iax.conf

 The idea being each user has a subdirectory containing their user-specific
 config data, so it would usually contain either sip.conf or iax.conf,
 always extensions.conf, and possibly dundi.conf

 Will this method no longer work in 1.6 upwards? I'm sure there must be
 others who've taken a similar approach to ensuring user portability between
 servers (moving users is as simple as grabbing the user-specific directory
 and dropping it on a different server). Do I need to ensure there's an
 empty file for each include in each directory?

Yes, an empty file would do the trick, but I think you only need one in a 
single directory for the glob not to return a warning.

The issue has been that in the past, Asterisk would silently ignore a file
which did not exist, which created problems for people who might have a simple
typo in their #include names and thus create nonworking dialplans.

If this would create a real problem for you, I'm sure we could find
alternative ways around this.  Perhaps a warning that a file does not exist
is enough of a warning that somebody can either ignore (at their own peril)
or fix.

-- 
Tilghman

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Re: [asterisk-users] iLBC codec

On Thursday 04 September 2008 05:55:48 Nhadie wrote:
 Hi,

 I am trying to test the ilbc codec on asterisk.

 allow=ilbc
 disallow=all

 using zoiper on two extensions, i set codec on zoiper to ilbc and
 disabled other codecs

 tested a call, looked at the channel:

NativeFormats: 0x400 (ilbc)
  WriteFormat: 0x40 (slin)
   ReadFormat: 0x40 (slin)
   WriteTranscode: Yes
ReadTranscode: Yes

 based on that it looks like it's transcoding  between ilbc and slin.
 isn't it supposed to use ilbc and not do transcoding?

If you're doing anything at all that requires Asterisk to uncompress the
audio (recording, mixing, conferencing, spying, etc.), then that would be the
reason.  You cannot directly mix a compressed codec; you have to decompress
the stream first.  Similarly, if you're recording, for example, a wav file,
one of the steps in recording that wav file is to decompress the audio stream.

-- 
Tilghman

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Re: [asterisk-users] G722 and Asterisk 1.6

2008-09-04 Thread Peder @ NetworkOblivion

 I'd also be more sold on it if it had half the features of the GXP2000 
 (which is only a little over half the price).

Sure, but if only half of the features in the GXP2000 actually work, 
what is the point of them?  I'd take a stable phone with less features 
over one that has lots of features that don't work correctly any day. 
I've opened numerous tickets with Grandstream and their answer is always 
we will look into it and they never reply, or that doesn't work and 
then no reply.

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[asterisk-users] strange transfer problem

2008-09-04 Thread Rizwan Hisham

Hi all,
I am having a strange problem with my asterisk server. When i dial an
outside tollfree number, if there is a menu for example press 1 for support,
press 2 for sales etc, after pressing any given option as the system begins
to transfer me the call hangs up. I have tried it so many times on different
tollfree numbers but the problem remains only when i dial from my asterisk
box. The call is disconnected with a normal call clearing (hangup cause 16).
But when i dial from my cell phone or any other line, the call is transfered
without any problem.

I also checked sip debug and core debug for different calls. I think it has
something to do with strict routing,  the only strange message i get on the
cli is:

[Sep  4 09:10:39] DEBUG[14929]: chan_sip.c:5690 reqprep: Strict routing
enforced for session [EMAIL PROTECTED]

This message does not appear for other calls (when there is no transfering)

I googled a little on strict and loose routing but i did not get it. maybe
someone here can help me solve this problem.

VERSIONS
asterisk 1.4.2
zaptel and libpri 1.4.0

I can send you core debug if you want it.

-- 
Best Regards
Rizwan Hisham
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Re: [asterisk-users] How to check mailbox exists (Received SIP subscribe for peer without mailbox)

2008-09-04 Thread Rob Hillis

Olivier wrote:
 Now that root cause is found, would you say that warnings or CLI 
 should have been different ?

 Obviously, MWI subscriptions must come from SIP hardphones (at least 
 those supporting MWI feature).
 So in this case, Received SIP subscribe for peer without mailbox: 
 9163 rather means Asterisk is receiving from SIP/9163, subscriptions 
 to MWI, but nothing in peer SIP/9163 settings is describing which 
 mailbox should be the scope of such subscriptions.

 Do you agree ?

Actually if you read the log entry, it can easily be interpreted that 
way.  Log entries really are a delicate balancing act - you need to 
provide enough information to determine what the problem is without 
becoming too wordy.  The only real way I could see to improve that log 
entry would be to say Received SIP subscribe for peer without 
*configured* mailbox: 9163  The log correctly identifies it as a 
notice since it's not an error condition.  Any log entry longer than 
this would bloat log files even further than they already are - and 
Asterisk log files on even a moderately busy system are already about as 
easy to follow as your average plate of spaghetti.


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[asterisk-users] ringback when the channel is answered

Hi guys,
I am trying to configure an asterisk server for our office.
Asterisk 1.4.17 SIP only

The problem appears when the call comes from external point to our
internal network. So when the server receives the call the channel is
answered and the remote user hears prompt which invite him to enter
internal private number. After that the server starts to wait the
extension. After timeout the server executes Dial application and
sends invite to sip client from our internal network. The problem is
in this point. I want to play ringback tone to remote user when he
waits internal user to pick up his phone but I could not instruct
Asterisk to generate fake ringback in rtp stream .

Is there a solution for this?

Thanks in advance.

-- 
Best Regards
eng. Anatoli Marinov

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Re: [asterisk-users] ringback when the channel is answered

2008-09-04 Thread Eric ManxPower Wieling

It will do so by default if you have a valid 
/etc/asterisk/indications.conf (only used for inband tones like after an 
Answer())

eng. Anatoli Marinov wrote:
 Hi guys,
 I am trying to configure an asterisk server for our office.
 Asterisk 1.4.17 SIP only
 
 The problem appears when the call comes from external point to our
 internal network. So when the server receives the call the channel is
 answered and the remote user hears prompt which invite him to enter
 internal private number. After that the server starts to wait the
 extension. After timeout the server executes Dial application and
 sends invite to sip client from our internal network. The problem is
 in this point. I want to play ringback tone to remote user when he
 waits internal user to pick up his phone but I could not instruct
 Asterisk to generate fake ringback in rtp stream .


-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] G722 and Asterisk 1.6

On Thu, Sep 4, 2008 at 7:57 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
 I'd also be more sold on it if it had half the features of the GXP2000
 (which is only a little over half the price).

 Sure, but if only half of the features in the GXP2000 actually work,
 what is the point of them?  I'd take a stable phone with less features
 over one that has lots of features that don't work correctly any day.
 I've opened numerous tickets with Grandstream and their answer is always
 we will look into it and they never reply, or that doesn't work and
 then no reply.


I agree! I bought a GXP1200 (business class phone) and it's buggy.
Can't use the message button (404 not found).. and some other features
(404 not found). I have requested help from Grandstream and so far
nothing.

I don't really think they test important features supported by their
phones just the basic ones (dial out/dial in) and that about it :)

Steve

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Re: [asterisk-users] ASTERISK supported Video phone

On Thu, Sep 4, 2008 at 1:48 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
 On Thu, 4 Sep 2008, Tharanga wrote:

 Hi folks,

 Can some one recommend a good video phone for asterisk (SIP.IAX2). I need
 better quality,  duarability. and should support various video codec's
 .(Codec auto negotiation support id prefferble)

 I suspect that the choices are so limited right now that good or bad is
 going to be very subjective. Grandstream GXP3000's appear to work from what
 I've heard others say, as does X-Lite (eyebeam?) softphone, Ekega
 (softphone) and ATL video phones...

 Then theres Polycoms with an extra zero aded to the price...

 Some people have reported good results with the BT Videophone 1000 units
 too.. (avalable for £60 a pair, but they need to have the early s/w release
 on them)

 I'm just about to order up a paid of Grandstreams for a project...

 (Hm. Can I trunk video over IAX?)


Dlink has launched one in india.
http://www.techgadgets.in/misc-gadgets/2008/13/d-link-gvc-3000-ip-videophone-and-glv-540-ip-phones-announced-in-india/

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Re: [asterisk-users] G722 and Asterisk 1.6

On Thu, 4 Sep 2008 21:12:52 +0530, Steve Repo wrote:

On Thu, Sep 4, 2008 at 7:57 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
 I'd also be more sold on it if it had half the features of the GXP2000
 (which is only a little over half the price).

 Sure, but if only half of the features in the GXP2000 actually work,
 what is the point of them?  I'd take a stable phone with less features
 over one that has lots of features that don't work correctly any day.
 I've opened numerous tickets with Grandstream and their answer is always
 we will look into it and they never reply, or that doesn't work and
 then no reply.


I agree! I bought a GXP1200 (business class phone) and it's buggy.
Can't use the message button (404 not found).. and some other features
(404 not found). I have requested help from Grandstream and so far
nothing.

I don't really think they test important features supported by their
phones just the basic ones (dial out/dial in) and that about it :)

Steve

Life is just too short to use a cheap phone. You can get good deals on
Polycom's and they are a pleasure to use, so that's what I stick with.

I also like Aastra  snom. Cisco pricing is simply offensive.

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] ringback when the channel is answered

Is there any special option which I should enable to activate these tones?
My progressinband is yes and I cal Dial app with r option it it right?



2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
 It will do so by default if you have a valid
 /etc/asterisk/indications.conf (only used for inband tones like after an
 Answer())

 eng. Anatoli Marinov wrote:
 Hi guys,
 I am trying to configure an asterisk server for our office.
 Asterisk 1.4.17 SIP only

 The problem appears when the call comes from external point to our
 internal network. So when the server receives the call the channel is
 answered and the remote user hears prompt which invite him to enter
 internal private number. After that the server starts to wait the
 extension. After timeout the server executes Dial application and
 sends invite to sip client from our internal network. The problem is
 in this point. I want to play ringback tone to remote user when he
 waits internal user to pick up his phone but I could not instruct
 Asterisk to generate fake ringback in rtp stream .


 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

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 Register Now: http://www.astricon.net

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-- 
Best Regards
eng. Anatoli Marinov

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Re: [asterisk-users] PRI Splitter

2008-09-04 Thread FailSafe Inc.

Everyone Interested,

The FSV-4PFS Order page has options for which pins you would like switched.

The default choice is T1/E1/POTS Pins (Pins 1,2,4,5)

Other possible choices are:
Ethernet (Pins 1,2,3,6)
and
All 8 Pins


Igor - you and I spoke before you ordered your devices.  I knew that you
intended to switch ethernet, so I of course picked that choice for you.

Craig - If you ordered the Ethernet switch and we mistakenly sent you a
T1/E1/Pots switch, please let me know via e-mail.  We'll get it exchanged
out right away.  Looking at your original order on our website, I don't see
the Ethernet option selected.  Either way, if you need a different
configuration, we will be happy to get it taken care of for you.

Bill
FailSafeVOIP, Inc.
Safe is always better than failed
http://www.failsafevoip.com
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


--
 Date: Thu, 04 Sep 2008 07:07:28 -0400
 From: Igor Hernandez [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] PRI Splitter
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=UTF-8

 Hy Craig,

 Can you elaborate on that? In our setup we have it doing just that and
 it works without a glitch.

 Regards,

 Igor H.

 Craig Guy wrote:
 The FSV-4PFS as shipped will not switch Ethernet ? it switches pins
 1,2,4,5.



 Craig



 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of *FailSafe
 Inc.
 *Sent:* Tuesday, 2 September 2008 11:27 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* Re: [asterisk-users] PRI Splitter



 Although the original topic of this thread has changed quite a bit, I
 wanted to point out that the SPF Product that you are discussing is
 quite similar to our product, the FSV-4PFS.  Ours is a 4 port device
 which can switch 4 T1/E1/J1/Ethernet or as many as 16 analog lines from
 a primary to a backup server.  It uses similar logic (power outage =
 failover server, loss of hearbeat = failover server) and also has a
 physical mechanical switch on the front of it which allows manual
 override switching to main or secondary server.



 We also have addressed the 'clean startup' that was discussed a few
 posts back.  The switch will start and remain in 'failover mode' until
 such time as it receives a hearbeat or the physical switch is moved to
 the main' position.  A failed main server can be restarted/repowered
 without bothering the backup server operation one bit - until you are
 ready to switch back to the main server.



 http://www.failsafevoip.com/index.php?main_page=product_infoproducts_id=1
 http://www.failsafevoip.com/index.php?main_page=product_infoproducts_id=1




 --
 FailSafeVOIP, Inc.
 Safe is always better than failed
 http://www.failsafevoip.com
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]



 On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said:

 that when both servers power fail you have a problem no matter if the

 failover switch ist still working or not.



 You've got that right my friend! :-)



 On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said:

 http://store.variantdistribution.com/category-s/49.htmVariant - one of

 Rhinos distributors and the only source I was able to find

 - quotes the card for US$ 700.



 Strange.  I've seen this happen before where retailers will list

 outrageously high prices for soon-to-be-released products.   For example

 the SNOM KlarVoice handset.  MSRP is $32, but I've seen it advertised
 for $200!



 http://www.8774e4voip.com/SearchResults.asp?Search=klarvoice



 I can say with confidence that the LIST price is US $350.  The street
 price will be considerably lower.  Frankly, if I were Snom or Rhino I'd
 be pretty cheezed off about this phenomenon.  After hearing the 'buzz'

 about a new product such as this, I'd hate for customers to *decide*
 against it mistkenly believing this incorrect price.  I'd turn my nose
 at either of these two products for the incorrect prices I've seen
 advertised.



 We're pretty stoked to have stumbled onto this product because it's
 brand new, and we've been looking for something like it for some time.



 -Karl


 

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Re: [asterisk-users] ASTERISK supported Video phone

I use it with n800 device - nokia internet tablet and standard nokia
soft phone I have video call. The codec that I use is h263 and it
works great.

2008/9/4 Steve Repo [EMAIL PROTECTED]:
On Thu, Sep 4, 2008 at 1:48 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Thu, 4 Sep 2008, Tharanga wrote:

Hi folks,

Can some one recommend a good video phone for asterisk (SIP.IAX2). I need
better quality, duarability. and should support various video codec's
.(Codec auto negotiation support id prefferble)

I suspect that the choices are so limited right now that good or bad is
going to be very subjective. Grandstream GXP3000's appear to work from what
I've heard others say, as does X-Lite (eyebeam?) softphone, Ekega
(softphone) and ATL video phones...

Then theres Polycoms with an extra zero aded to the price...

Some people have reported good results with the BT Videophone 1000 units
too.. (avalable for £60 a pair, but they need to have the early s/w release
on them)

I'm just about to order up a paid of Grandstreams for a project...

(Hm. Can I trunk video over IAX?)

Dlink has launched one in india.
http://www.techgadgets.in/misc-gadgets/2008/13/d-link-gvc-3000-ip-videophone-and-glv-540-ip-phones-announced-in-india/

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--
Best Regards
eng. Anatoli Marinov

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[asterisk-users] extensions.conf programming?

2008-09-04 Thread Ken D'Ambrosio

Hey, all.  I haven't really gotten deep into Asterisk since 1.0.x, and I'm
afraid I've forgotten a fair bit.  One big thing that I've forgotten is
the syntax, etc., for extensions.conf.  Where do I find that?  I'm looking
for stuff about commands, syntax for commands, variables, etc.  Is there a
handy-dandy manpage, webpage, or what-have-you that I'm missing?

Thanks!

-Ken


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Re: [asterisk-users] G722 and Asterisk 1.6

2008-09-04 Thread Thomas Kenyon

Steve Repo wrote:
 
 
 I agree! I bought a GXP1200 (business class phone) and it's buggy.
 Can't use the message button (404 not found).. and some other features
 (404 not found). I have requested help from Grandstream and so far
 nothing.
 
I've never heard of that problem, ar eyou sure the 404 response isn't 
coming from asterisk? (It works on all the GXP-2000's I have).

The Lines all work, the BLF keys mostly work (It's better than not 
having any), The Transfer, Conference, Message (can have a separate 
mailbox setup for each line) etc. buttons all work.

It's a shame they don't have a more standard headset port (like the 
Polycom). I havemn't experienced any problems with crashes or sound 
quality. (Although as I stated before, the G.722 codec isn't discernably 
clearer than the G.711 ones).

 I don't really think they test important features supported by their
 phones just the basic ones (dial out/dial in) and that about it :)
 
Apparently there are problems with some hardware versions, (I have 3 
different versions that have been okay) and most versions of the firmware.

I'm on 1.1.6.16 and it's been fine. Don't see why it is worth spending 
almost twice the money to get a handset without features that (in the 
case of BLF) are needed.

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[asterisk-users] 1.6rc4 chan_iax2 messages

2008-09-04 Thread MFH

As a result of:  
http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html

I am seeing these messages when upgrading for 1.6b9 to 1.6rc4.  Is there 
something I should be doing to address this warning?

[Sep  4 13:31:34] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 14670, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 13:34:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 13447, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 13:39:14] WARNING[2956]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 14442, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 13:40:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 11096, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 13:42:14] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 14517, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 13:42:54] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 2761, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 13:43:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 10117, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 13:44:54] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 4215, but no such call exists (and 
I cannot remove lagid, either).

MARK.

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Re: [asterisk-users] ringback when the channel is answered

2008-09-04 Thread Eric ManxPower Wieling

This has nothing to do with the progressinband setting and you should 
never use the r option.

eng. Anatoli Marinov wrote:
 Is there any special option which I should enable to activate these tones?
 My progressinband is yes and I cal Dial app with r option it it right?
 
 
 
 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
 It will do so by default if you have a valid
 /etc/asterisk/indications.conf (only used for inband tones like after an
 Answer())

 eng. Anatoli Marinov wrote:
 Hi guys,
 I am trying to configure an asterisk server for our office.
 Asterisk 1.4.17 SIP only

 The problem appears when the call comes from external point to our
 internal network. So when the server receives the call the channel is
 answered and the remote user hears prompt which invite him to enter
 internal private number. After that the server starts to wait the
 extension. After timeout the server executes Dial application and
 sends invite to sip client from our internal network. The problem is
 in this point. I want to play ringback tone to remote user when he
 waits internal user to pick up his phone but I could not instruct
 Asterisk to generate fake ringback in rtp stream .

 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

 ___
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 Register Now: http://www.astricon.net

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   http://lists.digium.com/mailman/listinfo/asterisk-users

 
 
 

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] extensions.conf programming?

2008-09-04 Thread Mark Michelson

Ken D'Ambrosio wrote:
 Hey, all.  I haven't really gotten deep into Asterisk since 1.0.x, and I'm
 afraid I've forgotten a fair bit.  One big thing that I've forgotten is
 the syntax, etc., for extensions.conf.  Where do I find that?  I'm looking
 for stuff about commands, syntax for commands, variables, etc.  Is there a
 handy-dandy manpage, webpage, or what-have-you that I'm missing?
 
 Thanks!
 
 -Ken
 

Your best bet is to read chapters 5 and 6 of Asterisk: The Future of Telephony.

Here's a link for the book itself:
http://www.oreilly.com/catalog/9780596510480/

Here's a link for the downloadable pdf:
http://downloads.oreilly.com/books/9780596510480.pdf

Here's a link for the book in html format
http://tfot.leifmadsen.com

Best of luck to you!
Mark Michelson

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Re: [asterisk-users] ASTERISK supported Video phone

2008-09-04 Thread Gordon Henderson

On Thu, 4 Sep 2008, Steve Repo wrote:

 Dlink has launched one in india.
 http://www.techgadgets.in/misc-gadgets/2008/13/d-link-gvc-3000-ip-videophone-and-glv-540-ip-phones-announced-in-india/

That's even uglier than the Grandstream ;-)

And why does it remind me of the microsoft un-natural keyboard... ???

But if it works at a consumer price ...

Gordon

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Re: [asterisk-users] extensions.conf programming?

2008-09-04 Thread Darren Sessions


A cheaper alternative would be the voip wiki.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf




_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Sep 4, 2008, at 12:13 PM, Mark Michelson wrote:


Ken D'Ambrosio wrote:
Hey, all.  I haven't really gotten deep into Asterisk since 1.0.x,  
and I'm
afraid I've forgotten a fair bit.  One big thing that I've  
forgotten is
the syntax, etc., for extensions.conf.  Where do I find that?  I'm  
looking
for stuff about commands, syntax for commands, variables, etc.  Is  
there a

handy-dandy manpage, webpage, or what-have-you that I'm missing?

Thanks!

-Ken



Your best bet is to read chapters 5 and 6 of Asterisk: The Future of  
Telephony.


Here's a link for the book itself:
http://www.oreilly.com/catalog/9780596510480/

Here's a link for the downloadable pdf:
http://downloads.oreilly.com/books/9780596510480.pdf

Here's a link for the book in html format
http://tfot.leifmadsen.com

Best of luck to you!
Mark Michelson

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Re: [asterisk-users] G722 and Asterisk 1.6

On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:

Asterisk should work fine with any phone that supports that codec.   
Personally, I have only used it with Polycom phones.  Also, again,  
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has  
full support.

Any plans to implement G.722.1 now that it's under a royalty free
license?

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] G722 and Asterisk 1.6

2008-09-04 Thread Gordon Henderson

On Thu, 4 Sep 2008, Thomas Kenyon wrote:

 Steve Repo wrote:

 I agree! I bought a GXP1200 (business class phone) and it's buggy.
 Can't use the message button (404 not found).. and some other features
 (404 not found). I have requested help from Grandstream and so far
 nothing.

 I've never heard of that problem, ar eyou sure the 404 response isn't
 coming from asterisk? (It works on all the GXP-2000's I have).

 The Lines all work, the BLF keys mostly work (It's better than not
 having any), The Transfer, Conference, Message (can have a separate
 mailbox setup for each line) etc. buttons all work.

 It's a shame they don't have a more standard headset port (like the
 Polycom). I havemn't experienced any problems with crashes or sound
 quality. (Although as I stated before, the G.722 codec isn't discernably
 clearer than the G.711 ones).

 I don't really think they test important features supported by their
 phones just the basic ones (dial out/dial in) and that about it :)

 Apparently there are problems with some hardware versions, (I have 3
 different versions that have been okay) and most versions of the firmware.

 I'm on 1.1.6.16 and it's been fine. Don't see why it is worth spending
 almost twice the money to get a handset without features that (in the
 case of BLF) are needed.

I'd more or less agree with this - I've deployed dozens of GXP2000's. 
(So-far only half a dozen GXP1200's) There does seem to be an issue with 
old hardware though, but the current crop have been very stable under 
1.1.6.16. Everything I've tried works - even BLF's. Downloading phone 
books is a doddle and provisioning is easy with gsutil and a few bits I've 
written myself.

I've yet to find anything that comes close, price wise, and right now, 
price is what people (in the UK!) are whinging about...

Gordon

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Re: [asterisk-users] 1.6rc4 chan_iax2 messages

On Thursday 04 September 2008 12:59:33 MFH wrote:
 As a result of:
 http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html

 I am seeing these messages when upgrading for 1.6b9 to 1.6rc4.  Is there
 something I should be doing to address this warning?

 [Sep  4 13:31:34] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 14670, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:34:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 13447, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:39:14] WARNING[2956]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 14442, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:40:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 11096, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:42:14] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 14517, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:42:54] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 2761, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:43:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 10117, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:44:54] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 4215, but no such call exists (and
 I cannot remove lagid, either).

They're actually pretty harmless messages.  I may wind up moving them to
DEBUG only.  You'll only get them when a host is unreachable during a call,
though, so you may want to figure out why that host became unreachable.

-- 
Tilghman

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Re: [asterisk-users] 1.6rc4 chan_iax2 messages

2008-09-04 Thread MFH

I was on the call at the time and was not experiencing any apparent 
problems. 

As I was responding I did some further investigation and saw the 
messages even when there wasn't an active call (so I thought).  I looked 
at the active IAX channels:

[Sep  4 14:27:14] WARNING[2956]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 6467, but no such call exists (and 
I cannot remove lagid, either).
[Sep  4 14:35:14] WARNING[2958]: chan_iax2.c:1200 __send_lagrq: I was 
supposed to send a LAGRQ with callno 5345, but no such call exists (and 
I cannot remove lagid, either).

asterisk*CLI iax2 show channels
Channel   Peer UsernameID (Lo/Rem)  Seq 
(Tx/Rx)  Lag  Jitter  JitBuf  Format
(None)67.202.54.166(None)  02713/0  
1/0  0ms  -0001ms  ms  unknow
1 active IAX channel

and determined that this is a connection to an Amazon EC2 image that I 
was using for testing but recently deactivated:

asterisk*CLI iax2 show peers
Name/UsernameHost Mask Port  Status
asterisk2/aster  67.202.54.166   (S)  255.255.255.255  4569  
UNREACHABLE


[asterisk2]
   type=friend
   username=asterisk2
   secret=pass
   auth=md5
   host=asterisk2.myhost.com
   context=frompeer
   peercontext=frompeer
   qualify=yes
   trunk=yes

I'm going to remove this definition and see if the messages go away.

MARK.

Tilghman Lesher wrote:
 On Thursday 04 September 2008 12:59:33 MFH wrote:
   
 As a result of:
 http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html

 I am seeing these messages when upgrading for 1.6b9 to 1.6rc4.  Is there
 something I should be doing to address this warning?

 [Sep  4 13:31:34] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 14670, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:34:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 13447, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:39:14] WARNING[2956]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 14442, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:40:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 11096, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:42:14] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 14517, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:42:54] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 2761, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:43:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 10117, but no such call exists (and
 I cannot remove lagid, either).
 [Sep  4 13:44:54] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was
 supposed to send a LAGRQ with callno 4215, but no such call exists (and
 I cannot remove lagid, either).
 

 They're actually pretty harmless messages.  I may wind up moving them to
 DEBUG only.  You'll only get them when a host is unreachable during a call,
 though, so you may want to figure out why that host became unreachable.

   

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Re: [asterisk-users] ringback when the channel is answered

So as I understand the only thing that I can do is to set up
indications.conf. Ok I will try it tomorrow and will write again with
my results.

Thanks a lot.



2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
 This has nothing to do with the progressinband setting and you should
 never use the r option.

 eng. Anatoli Marinov wrote:
 Is there any special option which I should enable to activate these tones?
 My progressinband is yes and I cal Dial app with r option it it right?



 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
 It will do so by default if you have a valid
 /etc/asterisk/indications.conf (only used for inband tones like after an
 Answer())

 eng. Anatoli Marinov wrote:
 Hi guys,
 I am trying to configure an asterisk server for our office.
 Asterisk 1.4.17 SIP only

 The problem appears when the call comes from external point to our
 internal network. So when the server receives the call the channel is
 answered and the remote user hears prompt which invite him to enter
 internal private number. After that the server starts to wait the
 extension. After timeout the server executes Dial application and
 sends invite to sip client from our internal network. The problem is
 in this point. I want to play ringback tone to remote user when he
 waits internal user to pick up his phone but I could not instruct
 Asterisk to generate fake ringback in rtp stream .

 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

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 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

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-- 
Best Regards
eng. Anatoli Marinov

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[asterisk-users] DAHDI FAQ not up. Anyplace else?

2008-09-04 Thread sean darcy

http://www.asterisk.org/zaptel-to-dahdi is empty. Is there anyplace else 
   besides the README's and Upgrade.txt's for config info on updating?

sean


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Re: [asterisk-users] ringback when the channel is answered

2008-09-04 Thread Steve Totaro

Why is it an option if it should never be used?.

Thanks,
Steve Totaro

On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED] 
wrote:
 This has nothing to do with the progressinband setting and you should
 never use the r option.

 eng. Anatoli Marinov wrote:
 Is there any special option which I should enable to activate these tones?
 My progressinband is yes and I cal Dial app with r option it it right?



 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
 It will do so by default if you have a valid
 /etc/asterisk/indications.conf (only used for inband tones like after an
 Answer())

 eng. Anatoli Marinov wrote:
 Hi guys,
 I am trying to configure an asterisk server for our office.
 Asterisk 1.4.17 SIP only

 The problem appears when the call comes from external point to our
 internal network. So when the server receives the call the channel is
 answered and the remote user hears prompt which invite him to enter
 internal private number. After that the server starts to wait the
 extension. After timeout the server executes Dial application and
 sends invite to sip client from our internal network. The problem is
 in this point. I want to play ringback tone to remote user when he
 waits internal user to pick up his phone but I could not instruct
 Asterisk to generate fake ringback in rtp stream .

 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

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 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

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[asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination

2008-09-04 Thread Shaun Wingrin

The setup is as follows: SIP phone registers via international link to Asterisk 
Box 1 and calls mean't for termination on Asterisk Box 2 via Zaptel Channels 
need to be hairpinned from Box 1 to 2. How is sip.conf configured on Box 1 and 
2 so that we don't get an error: Failed to authenticate user when 1's 
extensions.conf uses SIP to dial Asterisk Box 2 . How do we ensure that RTP 
traffic flows from SIP phone registering at 1 directly to 2 without first 
passing through 2?

Tx

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Re: [asterisk-users] ringback when the channel is answered

I do not know but I could not set it up. :) bad luck maybe.


2008/9/4 Steve Totaro [EMAIL PROTECTED]:
 Why is it an option if it should never be used?.

 Thanks,
 Steve Totaro

 On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED] 
 wrote:
 This has nothing to do with the progressinband setting and you should
 never use the r option.

 eng. Anatoli Marinov wrote:
 Is there any special option which I should enable to activate these tones?
 My progressinband is yes and I cal Dial app with r option it it right?



 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
 It will do so by default if you have a valid
 /etc/asterisk/indications.conf (only used for inband tones like after an
 Answer())

 eng. Anatoli Marinov wrote:
 Hi guys,
 I am trying to configure an asterisk server for our office.
 Asterisk 1.4.17 SIP only

 The problem appears when the call comes from external point to our
 internal network. So when the server receives the call the channel is
 answered and the remote user hears prompt which invite him to enter
 internal private number. After that the server starts to wait the
 extension. After timeout the server executes Dial application and
 sends invite to sip client from our internal network. The problem is
 in this point. I want to play ringback tone to remote user when he
 waits internal user to pick up his phone but I could not instruct
 Asterisk to generate fake ringback in rtp stream .

 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

 ___
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 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

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-- 
Best Regards
eng. Anatoli Marinov

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Re: [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination

2008-09-04 Thread Anthony Francis

Shaun Wingrin wrote:
 The setup is as follows: SIP phone registers via international link 
 to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 
 via Zaptel Channels need to be hairpinned from Box 1 to 2. How is 
 sip.conf configured on Box 1 and 2 so that we don't get an error: 
 Failed to authenticate user when 1's extensions.conf uses SIP to 
 dial Asterisk Box 2 . How do we ensure that RTP traffic flows from SIP 
 phone registering at 1 directly to 2 without first passing through 2?
  
 Tx

 Shaun
 

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This happens through a sip re-invite, the problem you seem to be having 
is that box 1 is not authenticated to send calls to box 2.

Anthony

/Everything should be as simple as possible, but no simpler - Albert 
Einstien/

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Re: [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination

2008-09-04 Thread Terry Wilson

 The setup is as follows: SIP phone registers via international link  
 to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2  
 via Zaptel Channels need to be hairpinned from Box 1 to 2. How is  
 sip.conf configured on Box 1 and 2 so that we don't get an error:  
 Failed to authenticate user when 1's extensions.conf uses SIP to  
 dial Asterisk Box 2 . How do we ensure that RTP traffic flows from  
 SIP phone registering at 1 directly to 2 without first passing  
 through 2?

I think if you set up a peer for Box 1 on Box 2, and set insecure=port  
on those peers, that it will not try to auth calls that are from your  
other asterisk box.  Of course, you'd have to make sure in your  
diaplan that you restricted access to those calls appropriately.  For  
the RTP, setting canreinvite=yes one peers that you want to be able to  
send media directly to each other should allow the RTP behavior you  
are looking for, but keep in mind that if there are any NATs between  
the phones, things can get messy in a hurry.

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Re: [asterisk-users] ASTERISK supported Video phone

2008-09-04 Thread bilal ghayyad

And he can use Vidoe with SIP?

As I know that SIP still does not support video.

Regards
Bilal


  

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Re: [asterisk-users] Asterisk supported Video phone

2008-09-04 Thread Philipp Kempgen

bilal ghayyad schrieb:
 And he can use Vidoe with SIP?
 
 As I know that SIP still does not support video.

Of course *SIP* supports video.  :-)
It's *Asterisk* which mainly supports voice.

   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] DAHDI FAQ not up. Anyplace else?

2008-09-04 Thread Kevin P. Fleming

sean darcy wrote:
 http://www.asterisk.org/zaptel-to-dahdi is empty. Is there anyplace else 
besides the README's and Upgrade.txt's for config info on updating?

No, the content that was supposed to be there was put into the
Zaptel-to-DAHDI.txt files in the Asterisk 1.4 and Asterisk 1.6 releases
instead; I'll update that page to include links to those files.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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[asterisk-users] Polycom BLF - multiple buddies

2008-09-04 Thread Robert McNaught

Hi,

Can anyone please comment on what the issue may be with this.  I am
trying to set up an Polycom IP601 with multiple buddy icons displaying
endpoint status.

I am using a polycom IP601, sip 2.2.2.0084

In the phone1.cfg file I set:

   attendant attendant.uri=4158149992 attendant.reg=1/

Using this, I get a valid SUBSCRIBE and NOTIFY from the sip server and
a buddy icon on screen.

For this value, I seem to only be able to set one value for this - I
tried duplicating the above tag with a different uri value.  I have
reg 1 which is the only registered account on the telephone.  The uri
is the username of reg 1 (4158149992).  I am finding that this only
allows me to monitor the local endpoint (which is pretty useless) - I
can only get one buddy icon.  I can set this to another account on the
server (eg 4158149991) which allows me to monitor 4158149991's status,
but only that status (it does not seem to use the directory entries
below like it should.

I am setting up a MAC-directory.xml file on the provisioning server
with the following:

?xml version=1.0 standalone=yes?^M
directory
item_list
item
lnConnery/ln
fnSean/fn
ct4158149991/ct
sd1/sd
bw1/bw
/item
item
lnLazenby/ln
fnGeorge/fn
ct4158149994/ct
sd2/sd
bw1/bw
/item

/item_list
/directory

I believe that this is what I need to enable more than one buddy icon?
 Can you please point me in the right direction.   Only the polycom
screen, I can only see 1 buddy icon despite having 2 speed dial
entries.

Can someone please point me in the right direction...

Thanks

Robert

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Re: [asterisk-users] DID number

2008-09-04 Thread Marcelo Freitas

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Re: [asterisk-users] G722 and Asterisk 1.6

2008-09-04 Thread Steve Underwood

Michael Graves wrote:
 On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:

   
 Asterisk should work fine with any phone that supports that codec.   
 Personally, I have only used it with Polycom phones.  Also, again,  
 Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has  
 full support.
 

 Any plans to implement G.722.1 now that it's under a royalty free
 license?

   
Its only G.722.1 Annex C (the 32k sample/second version) which is 
royalty free, and there seem to be some funky conditions in the 
licencing which may cause licence conflicts with other code. The spec in 
the main body of G.722.1 (the 16k samples/second version) is not royalty 
free, and that is the rather more interesting codec for most people. I 
doubt G.718 or G.719 will be in any way open.

On top of that, the only open source software developer I know to have 
contacted Polycom about a licence for G.722.1C has failed to get any 
response after multiple tries.

Regards,
Steve



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Re: [asterisk-users] DID number

2008-09-04 Thread Philipp Kempgen

Marcelo Freitas schrieb:
 

Please try again with a better mail client which is able to get
those MIME parts right.  :-P

   Philipp Kempgen

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Re: [asterisk-users] extensions.conf programming?

 A cheaper alternative would be the voip wiki.
 http://www.voip-info.org/tiki-
 index.php?page=Asterisk%20config%20extensions.conf

Unfortunately, as advised by other asterisk users,
http://www.voip-info.org  is sometimes really not that up-to-date.
However, that does not mean that we should give up on using and updating
http://www.voip-info.org because I think it is still the best voip
resource.

The best way is still to double check with the asterisk version that you
have installed by running CLI like below:

*CLI core show function
AGENTARRAYBASE64_DECODE
BASE64_ENCODEBLACKLISTCALLERID
[...]
VMCOUNT
*CLI core show application
AddQueueMember   ADSIProg AgentCallbackLogin

[...]
ZapScan  ZapSendKeypadFacility

*CLI
  -= Info about application 'WaitExten' =-

[Synopsis]
Waits for an extension to be entered

[Description]
  WaitExten([seconds][|options]): This application waits for the user to
enter
a new extension for a specified number of seconds.
  Note that the seconds can be passed with fractions of a second. For
example,
'1.5' will ask the application to wait for 1.5 seconds.
  Options:
m[(x)] - Provide music on hold to the caller while waiting for an
extension.
   Optionally, specify the class for music on hold within
parenthesis.

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Re: [asterisk-users] Polycom BLF - multiple buddies

 I believe that this is what I need to enable more than one buddy icon?
  Can you please point me in the right direction.   Only the polycom
 screen, I can only see 1 buddy icon despite having 2 speed dial
 entries.


I have been able to successfully turned on presence (which is the term
used outside Polycom) on IP601.
As I can recall, you need to a) configure sip.conf in the [general] and
per [extn] context; b) code hint extn in extensions.conf c) turned on
presence on the phone which will be buddy watching others d) turn on bw
on the phone which I saw you did.

However, I have never set what you did as in below and have no idea what
they are.
 In the phone1.cfg file I set:
attendant attendant.uri=4158149992 attendant.reg=1/

Just check out voip wiki and there are useful information over there
about presence (but may not be that much about Polycom phones sadly :-(.

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Re: [asterisk-users] iLBC codec

2008-09-04 Thread ronald

Hi Sir,

For this call i did not do anything except just call the extension

exten = 100,1,Dial(SIP/100|20|t|M(setmusiconhold,moh-100))

that's how i dial the extension, does musiconhold make asterisk 
uncompress? but during the call i did not use music on hold. whereelse 
should i look at?


Regards
Ron

Tilghman Lesher wrote:
 
 If you're doing anything at all that requires Asterisk to uncompress the
 audio (recording, mixing, conferencing, spying, etc.), then that would be the
 reason.  You cannot directly mix a compressed codec; you have to decompress
 the stream first.  Similarly, if you're recording, for example, a wav file,
 one of the steps in recording that wav file is to decompress the audio stream.
 

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[asterisk-users] The question about the M(X)option of Dial

2008-09-04 Thread larry

Hello!

 I have a question of the M(X) option of the Dial, In this M(X), the X
represented the Macro which could be run. Would you tell me that could it
run more than one Macro in this option? And How to do it ?

Thanks

 Larry

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[asterisk-users] libpri 1.4.5 priindication

2008-09-04 Thread Jorge Mendoza

Good morning,

Into the libpri 1.4.5 announcement, it is stated that This version of
libpri retains the ability to operate in this mode, but it is now a
configurable option which defaults to being 'off'. The next releases of
Asterisk will have configuration options to turn this behaviour on if
the user desires

Is this related to priindication?
How I can to turn this option to on ? Which is the next release of
Asterisk?

Thanks

Jorge Mendoza

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Re: [asterisk-users] Asterisk supported Video phone

2008-09-04 Thread bilal ghayyad

No plan for Asterisk to support video?
What kind of benifit I can get when I have video phone registers with Asterisk?

Regards
Bilal


  

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[asterisk-users] Linksys 3102 - Call Waiting

2008-09-04 Thread Joseph

Is anybody using Call Waiting works on Linksys 3102? Does it work?

If I'm on the phone, I can hear a notification 'beep', but when I put first 
caller on hold the line is busy. 
Linksys registers to Asterisk. 

-- 
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Re: [asterisk-users] G722 and Asterisk 1.6

2008-09-04 Thread Darrick Hartman

Michael Graves wrote:
 On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:
 
 Asterisk should work fine with any phone that supports that codec.   
 Personally, I have only used it with Polycom phones.  Also, again,  
 Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has  
 full support.
 
 Any plans to implement G.722.1 now that it's under a royalty free
 license?

Michael,

Royalty free does not mean free.  I believe there still is an upfront 
cost that Polycom is charging.  Perhaps Digium can work out some sort of 
a deal now that Polycom recognizes Asterisk as a valid platform.

I'd definitely love to see it supported.  It's a great way to actually 
show some improvement in voice quality compared to the 100 year old 
copper technology that's in use today.

Darrick

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Re: [asterisk-users] Polycom BLF - multiple buddies

2008-09-04 Thread James Sneeringer

I'm using a similar feature on 550 and 650 phones, also running 2.2.2.
I've never used the attendant option to do it, though, so I'm not
sure how it differs from what I'm doing. Instead, on the phones that
are allowed to do this, I have the following in their XML config. You
could just as easily enable it in phone1.cfg for all phones:

feature feature.1.name=presence feature.1.enabled=1/

Reboot then phone, and then when you add a new entry to your speed
dial directory (or edit an existing one), you will see a new Watch
Buddy option, which corresponds to the bw line in the
MAC-directory.xml file. The speed dial icon changes from the
multiple-dots icon to a silhouette of a person or will blink when the
phone is not registered, and the LED will go red when they're on a
call. It still functions as a speed dial, too.

John Lee was also correct that Polycom needs Asterisk's help. In
extensions.conf (or .ael), you need to set a hint for any extension
you want your 501 to see. In sip.conf, you need to set allowsubscribe
to yes, and set subscribecontext to a context that can see those
extensions.

I'm using this on our attendant phone, which is a 650 with three
expansion modules. The phone is programmed with several dozen employee
extensions, with Buddy Watch enabled for all. This lets the
receptionist see who is on the phone, so callers she transfers aren't
surprised when they go to voicemail. It's not perfect, because it
doesn't display DND or queue login/pause status, but it's better than
nothing.

-James


On Thu, Sep 4, 2008 at 7:09 PM, Robert McNaught [EMAIL PROTECTED] wrote:
 I am using a polycom IP601, sip 2.2.2.0084
 In the phone1.cfg file I set:

   attendant attendant.uri=4158149992 attendant.reg=1/

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Re: [asterisk-users] Polycom BLF - multiple buddies

 It's not perfect, because it
 doesn't display DND or queue login/pause status, but it's better than
 nothing.

James, on a different note, is it true that at this stage, we can never
get any queue login status/light on Polycom phone?

I posted a query a few days ago but I have got 0 reply.

Any thoughts?


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[asterisk-users] dahdi tdm400p: no luck

2008-09-04 Thread sean darcy

As best i could figure it out, I've installed dahdi and rc4.

My TDM400P doesn't answer fxo or fxs.

/etc/dahdi/system.conf:
loadzone   = us
defaultzone=us
fxoks=1,2
fxsks=4

/etc/asterisk/chan_dahdi.conf:

[house-phones]
context=internal  ; Uses the [internal] context in extensions.conf
signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel
dahdichan = 1  ; Telephone attached to port 1
dahdichan = 2  ; Telephone attached to port 2

[pstn]
context=incoming-pstn-line  ; Incoming calls go to [incoming-pstn-line] 
in extensions.conf
signalling=fxs_ks ; fxs_ks Use FXS signalling for an FXO channel
faxdetect=incoming
busydetect=yes
dahdichan = 4  ; PSTN attached to port 4

dmesg:

dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.0.0-rc3
ACPI: PCI Interrupt Link [APC1] enabled at IRQ 16
ACPI: PCI Interrupt :01:05.0[A] - Link [APC1] - GSI 16 (level, 
low) - IRQ 16
PCI: Setting latency timer of device :01:05.0 to 64
Freshmaker version: 73
Freshmaker passed register test
Clocksource tsc unstable (delta = -71426924 ns)
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Not installed
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules)
INFO-xpp: revision trunk-r6056 MAX_XPDS=64 (8*8)
INFO-xpp: FEATURE: without BRISTUFF support
INFO-xpp: FEATURE: with PROTOCOL_DEBUG
INFO-xpp: FEATURE: with sync_tick() from DAHDI
INFO-xpp_usb: revision trunk-r6056
usbcore: registered new interface driver xpp_usb
dahdi: Registered tone zone 0 (United States / North America)

  dahdi_scan
[1]
active=yes
alarms=OK
description=Wildcard TDM400P REV I Board 5
name=WCTDM/4
manufacturer=Digium
devicetype=Wildcard TDM400P REV I
location=PCI Bus 01 Slot 06
basechan=1
totchans=4
irq=16
type=analog
port=1,FXS
port=2,FXS
port=3,none
port=4,FXO

CLI dahdi show status
Description  Alarms  IRQbpviol CRC4 
Fra Codi Options  LBO
Wildcard TDM400P REV I Board 5   OK  0  0  0 
CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)

But if I dial in, no dial tone, nothing on the cli.
And:

*CLI dahdi show channels
Chan Extension  Context Language   MOH Interpret 
BlockedState
  pseudodefaultdefault 
In Service

Tried dahdi_genconf. No help.

Reverted now.

Any help appreciated.

seam


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Re: [asterisk-users] G722 and Asterisk 1.6

H...this bears making some calls to Polycom. They've been very good
to me recently. Very approachable. I think that they're really trying
to deal better with the Asterisk community.

Michael

On Thu, 04 Sep 2008 21:42:12 -0500, Darrick Hartman wrote:

Michael Graves wrote:
 On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:
 
 Asterisk should work fine with any phone that supports that codec.   
 Personally, I have only used it with Polycom phones.  Also, again,  
 Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has  
 full support.
 
 Any plans to implement G.722.1 now that it's under a royalty free
 license?

Michael,

Royalty free does not mean free.  I believe there still is an upfront 
cost that Polycom is charging.  Perhaps Digium can work out some sort of 
a deal now that Polycom recognizes Asterisk as a valid platform.

I'd definitely love to see it supported.  It's a great way to actually 
show some improvement in voice quality compared to the 100 year old 
copper technology that's in use today.

Darrick

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--
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] Polycom BLF - multiple buddies

2008-09-04 Thread James Sneeringer

I believe DEVSTATE() in 1.6 (backported to 1.4 in various places) will
let you arbitrarily control BLF, so you could control it in the
dialplan when an agent logs in or out (or pauses, or whatever).

Separately, you might be able to use sipsak (http://sipsak.org/) to
construct a SIP message that essentially forges an event to cause a
BLF state change on the phone. This guy is using it to control the MWI
light, so maybe it could be modified to control BLF:

http://www.siliconvp.us/modules.php?name=Newsfile=articlesid=7

-James


On Thu, Sep 4, 2008 at 9:53 PM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
 It's not perfect, because it
 doesn't display DND or queue login/pause status, but it's better than
 nothing.

 James, on a different note, is it true that at this stage, we can never
 get any queue login status/light on Polycom phone?

 I posted a query a few days ago but I have got 0 reply.

 Any thoughts?


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Re: [asterisk-users] Polycom BLF - multiple buddies

2008-09-04 Thread James Sneeringer

Sorry, needed to add one more note. To clarify, my agent phones have a
speed dial assigned for their login, and another to pause/unpause. I
could then use DEVSTATE to enable or disable the light next to their
speed dial button based on their status. I can't use it to update
anything on the LCD screen.

Today I do it completely differently. I use the idle window
minibrowser, and each agent phone has its own page it loads. I wrote a
perl script that connects to the AMI to watch the status of our
agents, and for any status change, it updates this page to reflect
their status. Since Polycom doesn't let you push data out to the
phones, they have to poll on a regular interval. I think ours are set
to every 5 seconds. It's a hack and it's ugly, but it works.

-James


On Thu, Sep 4, 2008 at 10:38 PM, James Sneeringer [EMAIL PROTECTED] wrote:
 I believe DEVSTATE() in 1.6 (backported to 1.4 in various places) will
 let you arbitrarily control BLF, so you could control it in the
 dialplan when an agent logs in or out (or pauses, or whatever).

 Separately, you might be able to use sipsak (http://sipsak.org/) to
 construct a SIP message that essentially forges an event to cause a
 BLF state change on the phone. This guy is using it to control the MWI
 light, so maybe it could be modified to control BLF:

 http://www.siliconvp.us/modules.php?name=Newsfile=articlesid=7

 -James


 On Thu, Sep 4, 2008 at 9:53 PM, Lee, John (Sydney)
 [EMAIL PROTECTED] wrote:
 It's not perfect, because it
 doesn't display DND or queue login/pause status, but it's better than
 nothing.

 James, on a different note, is it true that at this stage, we can never
 get any queue login status/light on Polycom phone?

 I posted a query a few days ago but I have got 0 reply.

 Any thoughts?


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Re: [asterisk-users] Polycom BLF - multiple buddies

 Sorry, needed to add one more note. To clarify, my agent phones have a
 speed dial assigned for their login, and another to pause/unpause. I
 could then use DEVSTATE to enable or disable the light next to their
 speed dial button based on their status. I can't use it to update
 anything on the LCD screen.

James, very useful info especially about enable/disable the light next
to the speed dial button which is exactly what I am after.  I am
currently using 1.4.x and would be interested to know how this can be
achieved.


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Re: [asterisk-users] ASTERISK supported Video phone